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RecordRTC is a server-less (entire client-side) JavaScript library can be used to record WebRTC audio/video media streams. It supports cross-browser audio/video recording.
RecordRTC Documentation / RecordRTC Wiki Pages / RecordRTC Demo / WebRTC Experiments
RecordRTC is a JavaScript-based media-recording library for modern web-browsers (supporting WebRTC getUserMedia API). It is optimized for different devices and browsers to bring all client-side (pluginfree) recording solutions in single place.
Please check dev directory for development files.
Browser | Support |
---|---|
Firefox | Stable / Aurora / Nightly |
Google Chrome | Stable / Canary / Beta / Dev |
Opera | Stable / NEXT |
Android | Chrome / Firefox / Opera |
Microsoft Edge | Edge |
Media File | Bitrate/Framerate | encoders | Framesize | additional info |
---|---|---|---|---|
Audio File (WAV) | 1411 kbps | pcm_s16le | 44100 Hz | stereo, s16 |
Video File (WebM) | 60 kb/s | (whammy) vp8 codec yuv420p | -- | SAR 1:1 DAR 4:3, 1k tbr, 1k tbn, 1k tbc (default) |
npm install recordrtc
# you can use with "require" (browserify/nodejs)
var RecordRTC = require('recordrtc');
var recorder = RecordRTC(mediaStream, { type: 'audio'});
or using Bower:
bower install recordrtc
To use it:
<script src="./node_modules/recordrtc/RecordRTC.js"></script>
<!-- or -->
<script src="//cdn.WebRTC-Experiment.com/RecordRTC.js"></script>
<!-- or -->
<script src="//www.WebRTC-Experiment.com/RecordRTC.js"></script>
There are some other NPM packages regarding RecordRTC:
<script src="https://cdn.rawgit.com/webrtc/adapter/master/adapter.js"></script>
<script>
function successCallback(stream) {
// RecordRTC usage goes here
}
function errorCallback(errror) {
// maybe another application is using the device
}
var mediaConstraints = { video: true, audio: true };
navigator.mediaDevices.getUserMedia(mediaConstraints).then(successCallback).catch(errorCallback);
</script>
You'll be recording both audio/video in single WebM container. Though you can edit RecordRTC.js to record in mp4.
var recordRTC;
function successCallback(stream) {
// RecordRTC usage goes here
recordRTC = RecordRTC(MediaStream);
recordRTC.startRecording();
}
function errorCallback(errror) {
// maybe another application is using the device
}
var mediaConstraints = { video: true, audio: true };
navigator.mediaDevices.getUserMedia(mediaConstraints).then(successCallback).catch(errorCallback);
btnStopRecording.onclick = function () {
recordRTC.stopRecording(function (audioVideoWebMURL) {
video.src = audioVideoWebMURL;
var recordedBlob = recordRTC.getBlob();
recordRTC.getDataURL(function(dataURL) { });
});
};
Demo: AudioVideo-on-Firefox.html
var recordRTC = RecordRTC(mediaStream);
recordRTC.startRecording();
recordRTC.stopRecording(function(audioURL) {
audio.src = audioURL;
var recordedBlob = recordRTC.getBlob();
recordRTC.getDataURL(function(dataURL) { });
});
Simply set volume=0
or muted=true
over <audio>
or <video>
element:
videoElement.muted = true;
audioElement.muted = true;
Otherwise, you can pass some media constraints:
function successCallback(stream) {
// RecordRTC usage goes here
}
function errorCallback(errror) {
// maybe another application is using the device
}
var mediaConstraints = {
audio: {
mandatory: {
echoCancellation: false,
googAutoGainControl: false,
googNoiseSuppression: false,
googHighpassFilter: false
},
optional: [{
googAudioMirroring: false
}]
},
};
navigator.mediaDevices.getUserMedia(mediaConstraints).then(successCallback).catch(errorCallback);
Everything is optional except type:'video'
:
var options = {
type: 'video'
};
var recordRTC = RecordRTC(mediaStream, options);
recordRTC.startRecording();
recordRTC.stopRecording(function(videoURL) {
video.src = videoURL;
var recordedBlob = recordRTC.getBlob();
recordRTC.getDataURL(function(dataURL) { });
});
onAudioProcessStarted
Useful to recover audio/video sync issues inside the browser:
recordAudio = RecordRTC( stream, {
onAudioProcessStarted: function( ) {
recordVideo.startRecording();
}
});
recordVideo = RecordRTC(stream, {
type: 'video'
});
recordAudio.startRecording();
onAudioProcessStarted
fixes shared/exclusive audio gap (a little bit). Because shared audio sometimes causes 100ms delay...
sometime about 400-to-500 ms delay.
Delay depends upon number of applications concurrently requesting same audio devices and CPU/Memory available.
Shared mode is the only mode currently available on 90% of windows systems especially on windows 7.
Everything is optional except type:'gif'
:
// you must "manually" link:
// https://cdn.webrtc-experiment.com/gif-recorder.js
var options = {
type: 'gif',
frameRate: 200,
quality: 10
};
var recordRTC = RecordRTC(mediaStream, options);
recordRTC.startRecording();
recordRTC.stopRecording(function(gifURL) {
mediaElement.src = gifURL;
});
You can say it: "HTML/Canvas Recording using RecordRTC"!
<script src="//cdn.WebRTC-Experiment.com/RecordRTC.js"></script>
<script src="//cdn.webrtc-experiment.com/screenshot.js"></script>
<div id="elementToShare" style="width:100%;height:100%;background:green;"></div>
<script>
var elementToShare = document.getElementById('elementToShare');
var recordRTC = RecordRTC(elementToShare, {
type: 'canvas'
});
recordRTC.startRecording();
recordRTC.stopRecording(function(videoURL) {
video.src = videoURL;
var recordedBlob = recordRTC.getBlob();
recordRTC.getDataURL(function(dataURL) { });
});
</script>
See a demo: /Canvas-Recording/
autoWriteToDisk
Using autoWriteToDisk
; you can suggest RecordRTC to auto-write to indexed-db as soon as you call stopRecording
method.
var recordRTC = RecordRTC(MediaStream, {
autoWriteToDisk: true
});
autoWriteToDisk
is helpful for single stream recording and writing to disk; however for MRecordRTC
; writeToDisk
is preferred one.
writeToDisk
You can write recorded blob to disk using writeToDisk
method:
recordRTC.stopRecording();
recordRTC.writeToDisk();
getFromDisk
You can get recorded blob from disk using getFromDisk
method:
// get all blobs from disk
RecordRTC.getFromDisk('all', function(dataURL, type) {
type == 'audio'
type == 'video'
type == 'gif'
});
// or get just single blob
RecordRTC.getFromDisk('audio', function(dataURL) {
// only audio blob is returned from disk!
});
For MRecordRTC; you can use word MRecordRTC
instead of RecordRTC
!
Another possible situation!
var recordRTC = RecordRTC(mediaStream);
recordRTC.startRecording();
recordRTC.stopRecording(function(audioURL) {
mediaElement.src = audioURL;
});
// "recordRTC" instance object to invoke "getFromDisk" method!
recordRTC.getFromDisk(function(dataURL) {
// audio blob is automaticlaly returned from disk!
});
In the above example; you can see that recordRTC
instance object is used instead of global RecordRTC
object.
var options = {
type: 'video',
video: {
width: 320,
height: 240
},
canvas: {
width: 320,
height: 240
}
};
pauseRecording
recordRTC.pauseRecording();
resumeRecording
recordRTC.resumeRecording();
getDataURL
recordRTC.getDataURL(function(dataURL) {
mediaElement.src = dataURL;
});
getBlob
blob = recordRTC.getBlob();
toURL
window.open( recordRTC.toURL() );
save
recordRTC.save();
bufferSize
Here is how to customize Buffer-Size for audio recording?
// From the spec: This value controls how frequently the audioprocess event is
// dispatched and how many sample-frames need to be processed each call.
// Lower values for buffer size will result in a lower (better) latency.
// Higher values will be necessary to avoid audio breakup and glitches
// bug: how to minimize wav size?
// workaround? obviously ffmpeg!
// The size of the buffer (in sample-frames) which needs to
// be processed each time onprocessaudio is called.
// Legal values are (256, 512, 1024, 2048, 4096, 8192, 16384).
var options = {
bufferSize: 16384
};
var recordRTC = RecordRTC(audioStream, options);
Following values are allowed for buffer-size:
// Legal values are (256, 512, 1024, 2048, 4096, 8192, 16384)
You can write like this:
var options = {
'buffer-size': 16384
};
sampleRate
Here is jow to customize Sample-Rate for audio recording?
// The sample rate (in sample-frames per second) at which the
// AudioContext handles audio. It is assumed that all AudioNodes
// in the context run at this rate. In making this assumption,
// sample-rate converters or "varispeed" processors are not supported
// in real-time processing.
// The sampleRate parameter describes the sample-rate of the
// linear PCM audio data in the buffer in sample-frames per second.
// An implementation must support sample-rates in at least
// the range 22050 to 96000.
var options = {
sampleRate: 96000
};
var recordRTC = RecordRTC(audioStream, options);
Values for sample-rate must be greater than or equal to 22050 and less than or equal to 96000.
You can write like this:
var options = {
'sample-rate': 16384
};
No WinXP SP2 based "Chrome" support. However, RecordRTC works on WinXP Service Pack 3.
RecordRTC uses WebAudio API for stereo-audio recording. AFAIK, WebAudio is not supported on android chrome releases, yet.
Audio recording fails for mono
audio. So, try stereo
audio only.
Do you know "RecordRTC" fails recording audio because following conditions fails:
--allow-file-access-from-files
)If you see this error message: Uncaught Error: SecurityError: DOM Exception 18
; it means that you're using HTTP
; whilst your webpage is loading worker file (i.e. audio-recorder.js
) from HTTPS
. Both files's (i.e. RecordRTC.js
and audio-recorder.js
) scheme MUST be same!
If you explorer chromium code; you'll see that some APIs can only be successfully called for WAV
files with stereo
audio.
Stereo audio is only supported for WAV files.
...still investigating the actual issue of failure with mono
audio.
Media Stream Recording API (MediaRecorder object) is being implemented by both Firefox and Chrome. RecordRTC is also using MediaRecorder API for Firefox (nightly).
RecordRTC is unable to record "mono" audio on chrome; however it seems that we can covert channels from "stereo" to "mono" using WebAudio API, though. MediaRecorder API's encoder only support 48k/16k mono audio channel (on Firefox Nightly).
http://recordrtc.org/ is a documentation webpage for RecordRTC.js. It is open-sourced in github and everyone can collaborate to improve documentation.
To contribute:
RecordRTC.js
file (aka latest.js
file)jsdoc
tool, you can generate documentation HTML pages from latest.js
filelatest.js
file for documentation.Steps to contribute:
latest.js
filelatest.js
file in the resulting recordrtc.org
directoryrecordrtc.org
directory and replace in RecordRTC
github clone's gh-pages
section# First step: install recordrtc.org template and javascript file
npm install recordrtc.org
# Second step: generate HTML files from template & latest.js file
cd .\node_modules\recordrtc.org
# This command generates HTML pages from latest.js file
node_modules\.bin\jsdoc node_modules\recordrtc\RecordRTC.js -d .\..\..\recordrtc.org node_modules\recordrtc\README.md -t template
Now you'll see a directory with name recordrtc.org
.
# This command runs index.html file
# You can use it to preview HTML pages (doc files)
.\..\..\recordrtc.org\index.html
Now, you should fork this repository:
And push/pull recordrtc.org
directory to gh-pages
.
latest.js
file?RecordRTC is using comments format from jsdoc:
E.g.
/**
* Description
* @summary Summary
* @typedef Hello
* @example
* var some = new Something();
*/
Example - stopRecording
method:
/**
* This method stops recording. It takes single "callback" argument. It is suggested to get blob or URI in the callback to make sure all encoders finished their jobs.
* @param {function} callback - This callback function is invoked after completion of all encoding jobs.
* @method
* @memberof RecordRTC
* @instance
* @example
* recordRTC.stopRecording(function(videoURL) {
* video.src = videoURL;
* });
* @todo Implement <code class="str">recordRTC.stopRecording().getDataURL(callback);</code>
*/
stopRecording: stopRecording,
RecordRTC.js is released under MIT licence . Copyright (c) Muaz Khan.
FAQs
RecordRTC is a server-less (entire client-side) JavaScript library that can be used to record WebRTC audio/video media streams. It supports cross-browser audio/video recording.
The npm package recordrtc receives a total of 110,877 weekly downloads. As such, recordrtc popularity was classified as popular.
We found that recordrtc demonstrated a not healthy version release cadence and project activity because the last version was released a year ago. It has 1 open source maintainer collaborating on the project.
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