lamejs
libdatachannel
lamejs | libdatachannel | |
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5 | 27 | |
782 | 1,569 | |
- | - | |
0.0 | 9.3 | |
6 months ago | 3 days ago | |
JavaScript | C++ | |
GNU General Public License v3.0 or later | Mozilla Public License 2.0 |
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lamejs
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Having trouble understanding Int32Array , Int8Array etc;
const channels = [ new Float32Array(441), new Float32Array(441), ]; for (let i = 0, j = 0, n = 1; i < int16.length; i++) { const int = int16[i]; // If the high bit is on, then it is a negative number, and actually counts backwards. const float = int >= 0x8000 ? -(0x10000 - int) / 0x8000 : int / 0x7fff; // Deinterleave channels[(n = ++n % 2)][!n ? j++ : j - 1] = float; } // https://github.com/zhuker/lamejs/commit/e18447fefc4b581e33a89bd6a51a4fbf1b3e1660 const left = channels.shift(); const right = channels.shift(); let leftChannel, rightChannel; if (this.mimeType.includes('mp3')) { const sampleBlockSize = 441; leftChannel = new Int32Array(left.length); rightChannel = new Int32Array(right.length); for (let i = 0; i < left.length; i++) { leftChannel[i] = left[i] < 0 ? left[i] * 32768 : left[i] * 32767; rightChannel[i] = right[i] < 0 ? right[i] * 32768 : right[i] * 32767; } }
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Need help with ytdl-core and ExpressJS
`` async importEncoder() { if (this.mimeType.includes('mp3')) { const { lamejs } = await import(${this.src.origin}/lame.min.js`); this.mp3encoder = new lamejs.Mp3Encoder(2, 44100, 256); this.mp3controller = void 0; return this.mp3stream = new ReadableStream({ start: (_) => { return (this.mp3controller = _); }, }); } // ... } // ... write: async ({ timestamp }) => { const int8 = new Int8Array(441 * 4); const { value, done } = await this.inputReader.read(); if (!done) { int8.set(new Int8Array(value)); } else { console.log({ done }); return this.audioWriter.closed; } const int16 = new Int16Array(int8.buffer); // https://stackoverflow.com/a/35248852 const channels = [ new Float32Array(441), new Float32Array(441), ]; for (let i = 0, j = 0, n = 1; i < int16.length; i++) { const int = int16[i]; // If the high bit is on, then it is a negative number, and actually counts backwards. const float = int >= 0x8000 ? -(0x10000 - int) / 0x8000 : int / 0x7fff; // Deinterleave channels[(n = ++n % 2)][!n ? j++ : j - 1] = float; } // https://github.com/zhuker/lamejs/commit/e18447fefc4b581e33a89bd6a51a4fbf1b3e1660 const left = channels.shift(); const right = channels.shift(); let leftChannel, rightChannel; if (this.mimeType.includes('mp3')) { const sampleBlockSize = 441; leftChannel = new Int32Array(left.length); rightChannel = new Int32Array(right.length); for (let i = 0; i < left.length; i++) { leftChannel[i] = left[i] < 0 ? left[i] * 32768 : left[i] * 32767; rightChannel[i] = right[i] < 0 ? right[i] * 32768 : right[i] * 32767; } } // ... if (this.mimeType.includes('mp3')) { // Finish writing MP3 const mp3buf = this.mp3encoder.flush(); if (mp3buf.length > 0) { this.mp3controller.enqueue(new Uint8Array(mp3buf)); this.mp3controller.close(); } this.resolve(new Response(this.mp3stream).arrayBuffer()); }
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Who uses node executable without using npm?
Read this https://github.com/zhuker/lamejs/blob/master/LICENSE.
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BeReadable - Online Multilingual Audio Transcription and Recorder
Lamejs
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Is it possible to load multiple files and export them as one mp3 file (ToneJS)
Hi, yes, and you don't even need to bring the whole Tone.js lib for that but if you are using it anyway then Tone.Offline is the right thing for this, you get the audiobuffer as output which can be converted to .wav blob. As for .mp3 https://github.com/zhuker/lamejs or any other Lame implementation is worth looking at.
libdatachannel
- VoRS: Vo(IP) Simple Alternative to Mumble
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Simplicity of IRC
You can use https://github.com/paullouisageneau/libdatachannel for your C/C++ integration needs. It's 10k lines. So the answer is 0. Its required dependencies (I assume this as they are git submodules in deps) are more than 100k lines, though, srtp support making the bulk of it. On my machine it took 11 seconds to compile it.
Irssi is 64k lines (plus its dependencies), so I guess that makes WebRTC complicated.
Can't argue that DCC isn't simple, but perhaps the protocol deviced decades ago is a bit too simple.
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OBS merges AV1 support for WebRTC
Most of the work happened in the libdatachannel! You can check out my PR here[0]
[0] https://github.com/paullouisageneau/libdatachannel/commit/a6...
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Pure C WebRTC
I am really excited about https://github.com/sepfy/libpeer. It has examples ready for ESP32 etc....
When working on KVS I wasn't familiar with the embedded space at all. I saw 'heavyweight' embedded where you were running on Linux. Then you had RTOS/No OS at all. I wasn't prepared for these devices at all. If we can make WebRTC work in the embedded space I think it will really accelerate what developers are able to build!
Remotely driven cars, security cameras, robots in hospitals that bring iPads to infectious patients etc... Creative people are building amazing things. The WebRTC/video space needs to work harder and support them :)
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I love how diverse the WebRTC space is now. Outside of this implementation you have plenty of other options!
* https://github.com/shinyoshiaki/werift-webrtc (Typescript)
* https://github.com/pion/webrtc (Golang)
* https://github.com/webrtc-rs/webrtc (Rust)
* https://github.com/algesten/str0m (Rust)
* hhttps://github.com/sepfy/libpeer (C/Embedded)
* https://webrtc.googlesource.com/src/ (C++)
* https://github.com/sipsorcery-org/sipsorcery (C#)
* https://github.com/paullouisageneau/libdatachannel (C++)
* https://github.com/elixir-webrtc (Elixir)
* https://github.com/aiortc/aiortc (Python)
* GStreamer’s webrtcbin (C)
See https://github.com/sipsorcery/webrtc-echoes for examples of some running against each other.
- WebRTC for the Curious
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Building WebRTC server implementation for Erlang
This is not true, there are actually multiple WebRTC implementations in different languages besides the reference library: aiortc (python), libdatachannel (C++), sipsorcery (C#),webrtc-rs (rust), werift (Typescript), and Amazon Kinesis (C)
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I spent two years building a desktop environment that runs in the browser, it's finally in beta!
Use any means necessary to transfer your data across devices. Could be IPFS, could be FTP, could be EventSource, WebSocket, WebTransport, Fetch, whatever. See https://github.com/guest271314/secure-file-transfer; offscreen-webrtc, https://github.com/paullouisageneau/libdatachannel.
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Client side Rest server?
I've successfully used libdatachannel to Web pages to connect native applications and stream data to the browser.
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Security Framework
Alternatively you can use your server as a signaling server for WebRTC (Insertable Streams ("Breakout Box"), or data channels https://github.com/paullouisageneau/libdatachannel), then users (peers) can exchange data themselves and you don't need to store anything, see True End-to-End Encryption with WebRTC Insertable Streams, A complete example for a WebRTC datachannel with manual signaling.
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Datachannel video streaming?
there is also a c++ library that can be used to open a data channel connection, I think a number of SFU “servers?” use this library (I wish I had) https://libdatachannel.org/
What are some alternatives?
vue-audio-visual - VueJS audio visualization components
libjuice - JUICE is a UDP Interactive Connectivity Establishment library
Vue.js - This is the repo for Vue 2. For Vue 3, go to https://github.com/vuejs/core
Pion WebRTC - Pure Go implementation of the WebRTC API
be-readable - BeReadable is a simple online tool that helps the user to turn audio into words and it can also record the audio and the user can play it or download it to mp3.
aiortc - WebRTC and ORTC implementation for Python using asyncio
banned - Banned from a site or organization? Account suspended? Censored? Why?
sora-unity-sdk - WebRTC SFU Sora Unity SDK
captureSystemAudio - Capture system audio ("What-U-Hear")
janus-gateway - Janus WebRTC Server
µWebSockets - Simple, secure & standards compliant web server for the most demanding of applications
webrtc - A pure Rust implementation of WebRTC