Menu
I have programmed a server that creates an audio stream from my MacBook's audio input, using express, osx-audio and lame:
const http = require("http");
const express = require("express");
const audio = require("osx-audio");
const lame = require("lame");
const audioInput = new audio.Input();
const encoder = new lame.Encoder({
channels: 2,
bitDepth: 16,
sampleRate: 44100,
bitRate: 128,
outSampleRate: 22050,
mode: lame.STEREO
});
audioInput.pipe(encoder);
const app = express();
const server = http.Server(app);
app.get("/stream.mp3", (req, res) => {
res.set({
"Content-Type": "audio/mpeg",
"Transfer-Encoding": "chunked"
});
encoder.pipe(res);
});
server.listen(3000);
On the client side, the sound from this audio stream is included as an <audio> element like so:
<audio controls autoplay preload="none">
<source src="./stream.mp3" type="audio/mpeg" />
<p>Oops – your browser doesn't support HTML5 audio!</p>
</audio>
This works – I can hear the sound from the input source I've selected on my laptop from any browser that is connected to the server when I click the “play” button on the audio element.
However the audio played by the browser lags several seconds behind the original signal. It seems although I'm using preload="none", the browser buffers quite a lot of the audio stream before it starts playing it.
Is there something obvious missing here? Is there a better way to achieve real-time audio with only a few milliseconds latency instead of several seconds?
If you're interested, my project's complete source code is here on GitHub.
749
To disable or enable low latency on your channel, head to your dashboard. Click the Hamburger icon, then open the Preferences drop down menu and select Stream. Latency mode settings can be found under the Stream Key & Preferences section at the top of the page. Here, you can pick Low latency or Normal latency.
For professional broadcasts, you want to aim for less than 15 seconds of latency. Dacast, for example, offers HLS delivery with RTMP ingests for low latency streaming. This reduces latency to 15 seconds or less. This is the type of latency that is acceptable for a professional broadcast.
With a lower latency, viewers are more likely to feel the issues between the encoder and the player. Network congestion and other factors may also cause live streaming issues, which can delay your stream. Delays can happen even when you have a great network that can sustain your average streaming bitrate.
To fix it, try: Improve your Internet connection (use ethernet, upgrade bandwidth, stop unnecessary network usage, or update your network device). Lower down the frame rate, bitrate, and resolution in your stream settings. Use dynamical changing bitrate in network settings.
Rather than reinventing the wheel, you can use FFMPEG, advertised as “A complete, cross-platform solution to record, convert and stream audio and video.”
Example:
ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
You can choose which library (h.264, mpeg, etc) your browser is OK with.
134
If you love us? You can donate to us via Paypal or buy me a coffee so we can maintain and grow! Thank you!
Donate Us With
