Open Source Apple iPhone Sound/Audio Apps

Namida is a modern, local-first music player that focuses on high-quality playback and robust library management. It scans device or desktop storage to build an organized library from folders and tags, presenting artists, albums, and tracks with artwork and rich metadata. The player emphasizes smooth navigation and playlist workflows, enabling favorites, custom lists, and quick queueing for flexible listening sessions. Audio features commonly include gapless playback and adjustable options aimed at uninterrupted, consistent sound. Its UI aims to be lightweight and visually coherent, keeping controls accessible without cluttering the screen. Namida’s architecture favors reliability and predictable performance across large libraries, minimizing lag as collections grow. The result is a straightforward but powerful tool for listeners who prioritize local files and responsive controls over cloud complexity.

Turn your phone or tablet into a fantastic audio viewer. Easy to use and customize. Choose from several preset settings of incredible colors or create your own. CAVA is a bar spectrum audio viewer based on my own open source project with the same name. Take the audio from the device's microphone and visualize the amplitude of the different frequencies as bars on the screen. Each bar represents a certain bandwidth of low to high frequencies. The leftmost bar starts at 50 Hz and the rightmost bar ends at 10 kHz. Although the frequencies outside this spectrum are audible, they do not contribute much to the overall sound image. You can also adjust the number of bars.

DPlayer is a lovely HTML5 danmaku video player to help people build video and danmaku easily. DPlayer is an MIT licensed open source project and completely free to use. However, the amount of effort needed to maintain and develop new features for the project is not sustainable without proper financial backing. Video events supported include abort, canplay, canplaythrough, durationchange, emptied, ended, error, loadeddata, loadedmetadata, loadstart, mozaudioavailable, pause, play, playing, progress, ratechange, seeked, seeking, stalled, suspend, timeupdate, volumechange, and waiting. Player events include screenshot, thumbnails_show, thumbnails_hide, danmaku_show, danmaku_hide, danmaku_clear, danmaku_loaded, danmaku_send, danmaku_opacity, contextmenu_show, contextmenu_hide, notice_show, notice_hide, quality_start, quality_end, destroy, resize, fullscreen, fullscreen_cancel, webfullscreen, webfullscreen_cancel, subtitle_show, subtitle_hide, and subtitle_change.

Google Play Music Desktop Player is open source on GitHub. This means YOU, the community, gets a say in all the features we implement and you can even get involved and help out. If you find a bug or want something new, tell us in Gitter or on GitHub and we will see what we can do! Being a lightweight, standalone framework we use FAR less resources than having Google Play Music open in a standard chrome tab. This frees up your computer's resources to do things you care about, instead of wasting them on playing music. Google Play Music Desktop Player adds a level of customization that simply isn't there in the web player. You can change your theme, customize the colors, send your play history straight to last.fm, and it even has a built-in equalizer. From advanced audio controls to simple song change notifications, this desktop player literally does it all.

AudioKit is an entire audio development ecosystem of code repositories, packages, libraries, algorithms, applications, playgorunds, tests, and scripts, built and used by a community of audio programmers, app developers, engineers, researchers, scientists, musicians, gamers, and people new to programming. An important goal for AudioKit is to allow it to grow and be maintainable by a handful of volunteers. For this reason we have extensive tests that are run whenever changes are made to any AudioKit code repository. We accept and encourage Github sponsorship of the people who spend a lot of time supporting AudioKit. We want to inspire the next generation of audio app developers and we do that by highlighting AudioKit-powered apps and by creating our own apps under the "AudioKit Pro" brand including the world's most downloaded synth "AudioKit Synth One" and a host of other AudioKit Pro apps.

​Supports microphone recording and real-time processing in most of the implemented getUserMediamobile and PC browsers, mainly including Chrome, Firefox, Safari, iOS 14.3+, Android WebView, Tencent Android X5 kernel (QQ, WeChat, Mini Program WebView) , uni-app (App, H5), and most Android phones updated after 2021 have their own browsers; do not support: UC-based kernel (typical Alipay), most of the old domestic mobile phones that have not been updated have their own browsers and any other form of browser (including PWA, WebClip, any App) on low-version iOS (11.0-14.2) except Safari inside page). Provides multiple plug-in function support. Rich audio visualization, variable speed and pitch processing, speech recognition, audio stream playback, etc.; with powerful real-time processing support, it can be used in various web applications: from simple recording to complex real-time voice Recognition (ASR), and even audio-related games, are handled with ease.

Modus is an open source, cross-platform C++ library which allows you to handle music from code. This means that you can: * Manage interactive and adaptive music * Use some kind of algorithm to improvise * Represent visually (simulate) musical performances * Select in real time the instruments that are going to play a previously written song * Let the user take part on the performance through any type of interface, by playing an instrument, changing the tempo, choosing the instruments, designing the structure of the song, etc. * Define song structures with metric modulations, accelerandos and ritardandos * Write your own scores, which can then be assigned to instruments to be played * Play along with a pre-recorded song or represent the performance * Everything else that comes into your head

Scan the network for your Media Server ("Server") and Renderer ("Renderer) devices, select one of each and then from the main screen select the music you want to play, drilling into Albums, Artists, Genres (all defined by your Media Server). Any music you want to hear is called a track and is loaded onto the play queue. Then select Play button to listen to the track listed at the top of the queue. You can also create playlists and store your favorite tracks for playback.

An iOS and macOS audio visualization framework built upon Core Audio useful for anyone doing real-time, low-latency audio processing and visualizations. I've designed six audio components and two interface components to allow you to immediately get your hands dirty recording, playing, and visualizing audio data. These components simply plug into each other and build on top of the high-performance, low-latency AudioUnits API and give you an easy to use API written in Objective-C instead of pure C. EZAudio was designed to work transparently across all iOS and OSX devices. This means one universal API whether you're building for Mac or iOS. For instance, under the hood an EZAudioPlot knows that it will subclass a UIView for iOS or an NSView for OSX and the EZMicrophone knows to build on top of the RemoteIO AudioUnit for iOS, but defaults to the system defaults for input and output for OSX.

Encodec is a neural audio codec developed by Meta for high-fidelity, low-bitrate audio compression using end-to-end deep learning. Unlike traditional codecs (like MP3 or Opus), Encodec uses a learned quantizer and decoder to reconstruct complex waveforms with remarkable accuracy at bitrates as low as 1.5 kbps. 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.

A binary waveform synthesizer derived from and based upon Timewave Zero theory by Terence McKenna. His theory proposes that the structure of time is a complex wave with a scalar potential, and the King Wen sequence of the I Ching is the 64 hexagram code from which the waveform of time is derived; that the King Wen sequence maps the linear progression of the human states of mind, which ultimately substantiates this waveform. This theory is the basis for the well-known novelty computer program he and others had developed. This program synth was made in supercollider.