RecordRTC: WebRTC 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.

Check all releases:

https://github.com/muaz-khan/RecordRTC/releases

Please check dev directory for development files.

Browsers Support:

Browser	Support	Features
Firefox	Stable / Aurora / Nightly	Audio+Video (Both local/remote)
Google Chrome	Stable / Canary / Beta / Dev	Audio+Video (Both local/remote)
Opera	Stable / NEXT	Audio/Vidoe Separately
Android	Chrome / Firefox / Opera	Audio/Vidoe Separately
Microsoft Edge	Normal Build	Only Audio

How RecordRTC encodes wav/webm?

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)

RecordRTC Demos

RecordRTC to Node.js
RecordRTC to PHP
RecordRTC to ASP.NET MVC
RecordRTC & HTML-2-Canvas i.e. Canvas/HTML Recording!
MRecordRTC i.e. Multi-RecordRTC!
RecordRTC on Ruby!
RecordRTC over Socket.io
ffmpeg-asm.js and RecordRTC! Audio/Video Merging & Transcoding!
RecordRTC / PHP / FFmpeg
Record Audio and upload to Nodejs server
ConcatenateBlobs.js - Concatenate multiple recordings in single Blob!
Remote audio-stream recording or a real p2p demo
Mp3 or Wav Recording

How to link?

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="https://cdn.WebRTC-Experiment.com/RecordRTC.js"></script>

<!-- or -->
<script src="https://www.WebRTC-Experiment.com/RecordRTC.js"></script>

It is suggested to link specific release:

https://github.com/muaz-khan/RecordRTC/releases

E.g.

<!-- use 5.2.6 or any other version -->
<script src="https://github.com/muaz-khan/RecordRTC/releases/download/5.2.8/RecordRTC.js"></script>

There are some other NPM packages regarding RecordRTC:

https://www.npmjs.org/search?q=RecordRTC

How to capture stream?

<script src="https://cdn.webrtc-experiment.com/gumadapter.js"></script>

<script>
function successCallback(stream) {
    // RecordRTC usage goes here
}

function errorCallback(error) {
    // maybe another application is using the device
}

var mediaConstraints = { video: true, audio: true };

navigator.mediaDevices.getUserMedia(mediaConstraints).then(successCallback).catch(errorCallback);
</script>

Record audio+video in Firefox

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

    var options = {
      mimeType: 'video/webm', // or video/mp4 or audio/ogg
      audioBitsPerSecond: 128000,
      videoBitsPerSecond: 128000,
      bitsPerSecond: 128000 // if this line is provided, skip above two
    };
    recordRTC = RecordRTC(stream, options);
    recordRTC.startRecording();
}

function errorCallback(error) {
    // 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

Record only Audio

var recordRTC = RecordRTC(mediaStream);
recordRTC.startRecording();
recordRTC.stopRecording(function(audioURL) {
    audio.src = audioURL;

    var recordedBlob = recordRTC.getBlob();
    recordRTC.getDataURL(function(dataURL) { });
});

Echo Issues

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(error) {
    // 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);

Constraints Reference

Record Video

Everything is optional except type:'video':

var options = {
   type: 'video',
   frameInterval: 20 // minimum time between pushing frames to Whammy (in milliseconds)
};
var recordRTC = RecordRTC(mediaStream, options);
recordRTC.startRecording();
recordRTC.stopRecording(function(videoURL) {
    video.src = videoURL;

    var recordedBlob = recordRTC.getBlob();
    recordRTC.getDataURL(function(dataURL) { });
});

Record animated GIF image

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 || canvas || context, options);
recordRTC.startRecording();
recordRTC.stopRecording(function(gifURL) {
   mediaElement.src = gifURL;
});

Record a Webpage

You can say it: "HTML/Canvas Recording using RecordRTC"!

<script src="https://cdn.WebRTC-Experiment.com/RecordRTC.js"></script>
<script src="https://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/

Record `<canvas>`

You can even record Canvas2D drawings:

<script src="https://cdn.webrtc-experiment.com/RecordRTC/Whammy.js"></script>
<script src="https://cdn.webrtc-experiment.com/RecordRTC/CanvasRecorder.js"></script>
<canvas></canvas>
<script>
var canvas = document.querySelector('canvas');
var recorder = new CanvasRecorder(window.canvasElementToBeRecorded, {
    disableLogs: false
});

// start recording <canvas> drawings
recorder.record();

// a few minutes later
recorder.stop(function(blob) {
    var url = URL.createObjectURL(blob);
    window.open(url);
});
</script>

Live Demo:

https://www.webrtc-experiment.com/RecordRTC/Canvas-Recording/record-canvas-drawings.html

Watch a video: https://vimeo.com/152119435

API Reference

`initRecorder`

It is a function that can be used to initiate recorder however skip getting recording outputs. It will provide maximum accuracy in the outputs after using startRecording method. Here is how to use it:

var audioRecorder = RecordRTC(mediaStream, {
  recorderType: StereoAudioRecorder
});

var videoRecorder = RecordRTC(mediaStream, {
  recorderType: WhammyRecorder
});

videoRecorder.initRecorder(function() {
  audioRecorder.initRecorder(function() {
    // Both recorders are ready to record things accurately
    videoRecorder.startRecording();
    audioRecorder.startRecording();
  });
});

After using stopRecording, you'll see that both WAV/WebM blobs are having following charachteristics:

Both are having same recording duration i.e. length
Video recorder is having no blank frames
Audio recorder is having no empty buffers

This method is really useful to sync audio/video outputs.

`setRecordingDuration`

You can ask RecordRTC to auto stop recording after specific duration. It accepts one mandatory and one optional argument:

recordRTC.setRecordingDuration(milliseconds, stoppedCallback);

// the easiest one:
recordRTC.setRecordingDuration(milliseconds).onRecordingStopped(stoppedCallback);

Try a simple demo; paste in the chrome console:

navigator.mediaDevices.getUserMedia({
    video: true
}).then(function(stream) {
    var recordRTC = RecordRTC(stream, {
        recorderType: WhammyRecorder
    });

    // auto stop recording after 5 seconds
    recordRTC.setRecordingDuration(5 * 1000).onRecordingStopped(function(url) {
        console.debug('setRecordingDuration', url);
        window.open(url);
    })

    recordRTC.startRecording();
}).catch(function(error) {
    console.error(error);
});

`clearRecordedData`

This method can be used to clear old recorded frames/buffers. Snippet:

recorder.clearRecordedData();

`recorderType`

If you're using recorderType then you don't need to use type. Second one will be redundant i.e. skipped.

You can force any Recorder by passing this object over RecordRTC constructor:

var audioRecorder = RecordRTC(mediaStream, {
  recorderType: StereoAudioRecorder
})

It means that ALL_BROWSERS will be using StereoAudioRecorder i.e. WebAudio API for audio recording.

This feature brings remote audio recording support in Firefox, and local audio recording support in Microsoft Edge.

Note: Chrome >=50 supports remote audio+video recording.

You can even force WhammyRecorder on Firefox however webp format isn't yet supported in standard Firefox builds. It simply means that, you're skipping MediaRecorder API in Firefox.

`type`

If you are NOT using recorderType parameter then type parameter can be used to ask RecordRTC choose best recorder-type for recording.

// if it is Firefox, then RecordRTC will be using MediaStreamRecorder.js
// if it is Chrome or Opera, then RecordRTC will be using WhammyRecorder.js
var recordVideo = RecordRTC(mediaStream, {
  type: 'video'
});

// if it is Firefox, then RecordRTC will be using MediaStreamRecorder.js
// if it is Chrome or Opera or Edge, then RecordRTC will be using StereoAudioRecorder.js
var recordVideo = RecordRTC(mediaStream, {
  type: 'audio'
});

`frameInterval`

Set minimum interval (in milliseconds) between each time we push a frame to Whammy recorder.

var whammyRecorder = RecordRTC(videoStream, {
  recorderType: WhammyRecorder,
  frameInterval: 1   // setTimeout interval
});

`disableLogs`

You can disable all the RecordRTC logs by passing this Boolean:

var recorder = RecordRTC(mediaStream, {
  disableLogs: true
});

`numberOfAudioChannels`

You can force StereoAudioRecorder to record single-audio-channel only. It allows you reduce WAV file size to half.

var audioRecorder = RecordRTC(audioStream, {
  recorderType: StereoAudioRecorder,
  numberOfAudioChannels: 1 // or leftChannel:true
});

It will reduce WAV size to half!

This feature is useful only in Chrome and Microsoft Edge (WAV-recorders). It can work in Firefox as well.

How to set video width/height?

var options = {
   type: 'video',
   video: {
      width: 320,
      height: 240
   },
   canvas: {
      width: 320,
      height: 240
   }
};

var recordVideo = RecordRTC(MediaStream, options);

`pauseRecording`

RecordRTC pauses recording buffers/frames.

recordRTC.pauseRecording();

`resumeRecording`

If you're using "initRecorder" then it asks RecordRTC that now its time to record buffers/frames. Otherwise, it asks RecordRTC to not only initialize recorder but also record buffers/frames.

recordRTC.resumeRecording();

`getDataURL`

Optionally get "DataURL" object instead of "Blob".

recordRTC.getDataURL(function(dataURL) {
   mediaElement.src = dataURL;
});

`getBlob`

Get "Blob" object. A blob object looks similar to input[type=file]. Which means that you can append it to FormData and upload to server using XMLHttpRequest object. Even socket.io nowadays supports blob-transmission.

blob = recordRTC.getBlob();

`toURL`

A virtual URL. It can be used only inside the same browser. You can't share it. It is just providing a preview of the recording.

window.open( recordRTC.toURL() );

`save`

Invoke save-as dialog. You can pass "fileName" as well; though fileName argument is optional.

recordRTC.save('File Name');

`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)

If you passed invalid value then you'll get blank audio.

`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.

If you passed invalid value then you'll get blank audio.

You can pass custom sample-rate values only on Mac (or additionally maybe on Windows 10).

`mimeType`

This option allows you set MediaRecorder output format (currently works only in Firefox; Chrome support coming soon):

var options = {
  mimeType 'video/webm', // or video/mp4 or audio/ogg
  bitsPerSecond: 128000
};
var recorder = RecordRTC(mediaStream, options);

Note: For chrome, it will simply auto-set type:audio or video parameters to keep supporting StereoAudioRecorder.js and WhammyRecorder.js.

That is, you can skip passing type:audio parameter when you're using mimeType parameter.

`bitsPerSecond`

The chosen bitrate for the audio and video components of the media. If this is specified along with one or the other of the above properties, this will be used for the one that isn't specified.

var options = {
  mimeType 'video/webm', // or video/mp4 or audio/ogg
  bitsPerSecond: 128000
};
var recorder = RecordRTC(mediaStream, options);

`audioBitsPerSecond`

The chosen bitrate for the audio component of the media.

var options = {
  mimeType 'audio/ogg',
  audioBitsPerSecond: 128000
};
var recorder = RecordRTC(mediaStream, options);

`videooBitsPerSecond`

The chosen bitrate for the video component of the media.

var options = {
  mimeType 'video/webm', // or video/mp4
  videooBitsPerSecond: 128000
};
var recorder = RecordRTC(mediaStream, options);

`onAudioProcessStarted`

Note: "initRecorder" is preferred over this old hack. Both works similarly.

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.

`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.

Credits

Recorderjs for audio recording
whammy for video recording
jsGif for gif recording

Spec & Reference

Contribute in RecordRTC.org domain

The domain www.RecordRTC.org is open-sourced here:

https://github.com/muaz-khan/RecordRTC/tree/gh-pages

License

RecordRTC.js is released under MIT licence . Copyright (c) Muaz Khan.

Name		Name	Last commit message	Last commit date
Latest commit History 95 Commits
Canvas-Recording		Canvas-Recording
MRecordRTC		MRecordRTC
PHP-and-FFmpeg		PHP-and-FFmpeg
RecordRTC-over-Socketio		RecordRTC-over-Socketio
RecordRTC-to-ASPNETMVC		RecordRTC-to-ASPNETMVC
RecordRTC-to-Nodejs		RecordRTC-to-Nodejs
RecordRTC-to-PHP		RecordRTC-to-PHP
dev		dev
libs		libs
.gitignore		.gitignore
.jshintrc		.jshintrc
.npmignore		.npmignore
.travis.yml		.travis.yml
AudioVideo-on-Firefox.html		AudioVideo-on-Firefox.html
Gruntfile.js		Gruntfile.js
README.md		README.md
Record-Mp3-or-Wav.html		Record-Mp3-or-Wav.html
RecordRTC.js		RecordRTC.js
RecordRTC.min.js		RecordRTC.min.js
bower.json		bower.json
index.html		index.html
package.json		package.json
server.js		server.js

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RecordRTC: WebRTC audio/video recording

Check all releases:

Browsers Support:

How RecordRTC encodes wav/webm?

RecordRTC Demos

How to link?

How to capture stream?

Record audio+video in Firefox

Record only Audio

Echo Issues

Record Video

Record animated GIF image

Record a Webpage

Record <canvas>

API Reference

initRecorder

setRecordingDuration

clearRecordedData

recorderType

type

frameInterval

disableLogs

numberOfAudioChannels

How to set video width/height?

pauseRecording

resumeRecording

getDataURL

getBlob

toURL

save

bufferSize

sampleRate

mimeType

bitsPerSecond

audioBitsPerSecond

videooBitsPerSecond

onAudioProcessStarted

autoWriteToDisk

writeToDisk

getFromDisk