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recordrtc

5.4.7 • Public • Published a month ago

RecordRTC: WebRTC JavaScript Library for Audio+Video+Screen Recording

Demo: https://www.webrtc-experiment.com/RecordRTC/

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+Video (Both local/remote)
Android	Chrome / Firefox / Opera	Audio+Video (Both local/remote)
Microsoft Edge	Normal Build	Only Audio - No Video - No Canvas - No Screen
Safari 11	preview/beta (OSX/iOS11)	Only StereoAudioRecorder - No Video - No Canvas - No Screen

Frameworks

Angular2 - check article and demo github repository - via #186
React.js - check this article and demo github repository
Video.js - check this github repository
meteor - check an old github repository

Want to add more? Please make a pull-request to update README.md

Tests?

https://travis-ci.org/muaz-khan/RecordRTC

Tests source code:

https://github.com/muaz-khan/RecordRTC/tree/master/test

Free?

It is MIT Licenced, which means that you can use it in any commercial/non-commercial product, free of cost.

RecordRTC Containers Format

RecordRTC supports vp9, vp8, h264, mkv, opus/vorbis, and pcm (mono/stereo).

vp9

var options = {
    recorderType: MediaStreamRecorder,
    mimeType: 'video/webm\;codecs=vp9'
};
var recordRTC = RecordRTC(stream, options);

vp8

var options = {
    recorderType: MediaStreamRecorder,
    mimeType: 'video/webm\;codecs=vp8'
};
var recordRTC = RecordRTC(stream, options);

h264

var options = {
    recorderType: MediaStreamRecorder,
    mimeType: 'video/webm\;codecs=h264'
};
var recordRTC = RecordRTC(stream, options);

pcm

var options = {
    recorderType: StereoAudioRecorder,
    mimeType: 'audio/wav'
};
var recordRTC = RecordRTC(stream, options);

opus

var options = {
    recorderType: MediaStreamRecorder,
    mimeType: 'audio/webm' // Firefox also supports: "audio/ogg"
};
var recordRTC = RecordRTC(stream, options);

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

You can also try a chrome extension for screen recording:

https://chrome.google.com/webstore/detail/recordrtc/ndcljioonkecdnaaihodjgiliohngojp

How to link?

NPM install

npm install recordrtc

# you can use with "require" (browserify/nodejs)
var RecordRTC = require('recordrtc');

var recorder = RecordRTC({}, {
    type: 'video',
    recorderType: RecordRTC.WhammyRecorder
});

console.log('\n--------\nRecordRTC\n--------\n');
console.log(recorder);

console.log('\n--------\nstartRecording\n--------\n');
recorder.startRecording();
console.log('\n--------\nprocess.exit()\n--------\n');

process.exit()

https://tonicdev.com/npm/recordrtc

Here is how to use require:

var RecordRTC = require('recordrtc');
var Whammy = RecordRTC.Whammy;
var WhammyRecorder = RecordRTC.WhammyRecorder;
var StereoAudioRecorder = RecordRTC.StereoAudioRecorder;
// and so on
 
var video = new Whammy.Video(100);
var recorder = new StereoAudioRecorder(stream, options);

<!-- link npm package scripts -->
<script src="./node_modules/recordrtc/RecordRTC.js"></script>

There are some other NPM packages regarding RecordRTC:

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

bower install

bower install recordrtc

<!-- link bower package scripts -->
<script src="./bower_components/recordrtc/RecordRTC.js"></script>

CDN

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

Otherwise check cdnjs below.

Releases

You can even link specific releases:

<!-- use 5.4.7 or any other version on cdnjs -->
<script src="https://cdnjs.cloudflare.com/ajax/libs/RecordRTC/5.4.7/RecordRTC.js"></script>

Note: You can use RecordRTC.min.js as well. (on webrtc-experiment or cdnjs)

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

You'll be recording both audio/video in single WebM or Mp4 container.

var recordRTC;
 
function successCallback(stream) {
    // RecordRTC usage goes here
 
    var options = {
      mimeType: 'video/webm', // or video/webm\;codecs=h264 or video/webm\;codecs=vp9
      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) { });
    });
};

Record only Audio

var recordRTC = RecordRTC(audioStream, { type: 'audio' });
recordRTC.startRecording();
recordRTC.stopRecording(function(audioURL) {
    audio.src = audioURL;
 
    var recordedBlob = recordRTC.getBlob();
    recordRTC.getDataURL(function(dataURL) { });
});

`options`

RecordRTC requires a second parameter named as options or configuration or hints or preferences:

var options = {
    recorderType: MediaStreamRecorder,
    mimeType: 'video/webm\;codecs=vp9'
};
var recordRTC = RecordRTC(stream, options);

You can pass options object over startRecording method as well:

var recordRTC = RecordRTC(stream);
 
var options = {
    recorderType: MediaStreamRecorder,
    mimeType: 'video/webm\;codecs=vp9'
};
recordRTC.startRecording(options);

type accepts video or audio or canvas or gif
mimeType accepts all these values
recorderType accepts MediaStreamRecorder or StereoAudioRecorder or WhammyRecorder or GifRecorder or any recorder type from this page
timeSlice accepts numbers in milliseconds; use this to force intervals-based blobs
ondataavailable pass this function along with timeSlice to get intervals based blobs
checkForInactiveTracks accepts true or false; use this to disable default inactive-stream-checker functions
onTimeStamp it is a function that is called-back by the MediaStreamRecorder; timeSlice parameter is required for this function
bitsPerSecond accepts numbers in bits; applies both to audio and video tracks
audioBitsPerSecond accepts numbers in bits; applies only to audio tracks
videoBitsPerSecond accepts numbers in bits; applies only to video tracks
disableLogs accepts true or false; use this to disable console logs
frameInterval accepts numbers in milliseconds; use this with MultiStreamRecorder, CanvasRecorder and WhammyRecorder
previewStream it is a function that is called-back by the MultiStreamRecorder
video accepts object similar to this: {width: 320, height: 240}; pass this parameter for MultiStreamRecorder, CanvasRecorder and WhammyRecorder
canvas accepts object similar to this: {width: 320, height: 240}; pass this parameter for MultiStreamRecorder, CanvasRecorder and WhammyRecorder
sampleRate used only by the StereoAudioRecorder
bufferSize used only by the StereoAudioRecorder
numberOfAudioChannels used only by the StereoAudioRecorder

Record Multiple Videos

Demos:

You can record many videos/streams in single WebM/Mp4 file (WebRTC Conference Recording):

var arrayOfStreams = [localStream, remoteStream1, remoteStream2, remoteStream3];
 
var recordRTC = RecordRTC(arrayOfStreams, {
  type: 'video',
  mimeType: 'video/webm', // or video/webm\;codecs=h264 or video/webm\;codecs=vp9
  previewStream: function(stream) {
    // it is optional
    // it allows you preview the recording video
  }
});
recordRTC.startRecording();
recordRTC.stopRecording(function(singleWebM) {
    video.src = singleWebM;
 
    var recordedBlob = recordRTC.getBlob();
    recordRTC.getDataURL(function(dataURL) { });
});

Points:

Instead of passing single MediaStream, you are passing array of MediaStreams
You will get single webm or mp4 according to your mimeType

MultiStreamRecorder.js supports two extra methods:

addStreams
resetVideoStreams

var msRecorder = recorder.getInternalRecorder();
if (msRecorder instanceof MultiStreamRecorder) {
    msRecorder.addStreams([newAudioStream]);
    msRecorder.resetVideoStreams([screenStream]);
}

Usecases:

You can add more audio and/or video streams during live recording (using addStreams method)
You can reset/remove/replace old videos using resetVideoStreams

resetVideoStreams can be used to recorded screenStream in full-screen mode e.g.

if (yourScreen.isScreen === true) {
    yourScreen.fullcanvas = true;
    yourScreen.width = window.screen.width;
    yourScreen.height = window.screen.height;
 
    // now it will record all audios + only_this_screen
    internalRecorder.resetVideoStreams([yourScreen]);
}

As soon as screen is stopped:

addStreamStopListener(yourScreen, function() {
    var cameraStreams = getSingleOrMultipleCameraStreams();
 
    // now it will record all audios + all_your_cameras
    internalRecorder.resetVideoStreams(cameraStreams);
});

`getInternalRecorder`

You can get access to internal recorders e.g. MultiStreamRecorder, MediaStreamRecorder, StereoAudioRecorder, WhammyRecorder etc.

Use getInternalRecorder only after startRecording. It may return NULL according to RecordRTC current state.

// if RecordRTC recording in-progress
if (recorder.state === 'recording') {
    // get MediaStreamRecorder
    var msRecorder = recorder.getInternalRecorder();
 
    // always check for NULL or verify the recorder type
    if (msRecorder instanceof MultiStreamRecorder) {
        // it is NOT NULL
        // also it is MultiStreamRecorder instance
        // now we can use these extra methods
        msRecorder.addStreams([newAudioStream]);
        msRecorder.resetVideoStreams([screenStream]);
    }
}

Internal recorders can add extra methods. Same as MultiStreamRecorder which is supporting two extra methods:

addStreams
resetVideoStreams

`onStateChanged`

Use this method to detect status of the recording:

recorder = RecordRTC(stream, {
    type: 'video',
    onStateChanged: function(state) {
        alert('Current recorder status: ' + state);
    }
});
 
recorder.startRecording();

`state`

Use this property to detect status of the recording:

recorder = RecordRTC(stream, {
    type: 'video'
});
 
alert('Current recorder status: ' + recorder.state);
 
recorder.startRecording();
 
alert('Current recorder status: ' + recorder.state);
 
recorder.stopRecording(function() {
    alert('Current recorder status: ' + recorder.state);
});

You can even use getState method:

alert('Current recorder status: ' + recorder.getState());

`version`

Detect current RecordRTC version:

recorder = RecordRTC(stream, {
    type: 'video'
});
 
alert('Current recorder version: ' + recorder.version);

You can even use RecordRTC.version:

alert('Current recorder version: ' + RecordRTC.version);

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`

Note: Firefox seems has a bug. It is unable to pause the recording. Seems internal bug because they correctly changes mediaRecorder.state from recording to paused but they are unable to pause internal recorders.

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 = {
   type: 'audio',
   recorderType: StereoAudioRecorder,
   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 how 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 = {
   type: 'audio',
   recorderType: StereoAudioRecorder,
   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).

`desiredSampRate`

Set sample rates such as 8K or 16K. Reference: http://stackoverflow.com/a/28977136/552182

// record 16khz audio
var options = {
   type: 'audio',
   recorderType: StereoAudioRecorder,
   desiredSampRate: 16 * 1000 // bits-per-sample * 1000
};
var recordRTC = RecordRTC(audioStream, options);

`mimeType`

This option allows you set MediaRecorder output format:

var options = {
  mimeType: 'video/webm', // or video/webm\;codecs=h264 or video/webm\;codecs=vp9
  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.

`isMimeTypeSupported`

function isMimeTypeSupported(mimeType) {
    // if (webrtcDetectedBrowser === 'edge') return false;
 
    if (typeof MediaRecorder.isTypeSupported !== 'function') {
        return true;
    }
 
    return MediaRecorder.isTypeSupported(mimeType);
}

Detect Audio Formats:

var mimeType = 'audio/mpeg';
var recorderType = MediaStreamRecorder;
 
if (isMimeTypeSupported(mimeType) === false) {
    console.log(mimeType, 'is not supported.');
    mimeType = 'audio/ogg';
 
    if (isMimeTypeSupported(mimeType) === false) {
        console.log(mimeType, 'is not supported.');
        mimeType = 'audio/webm';
 
        if (isMimeTypeSupported(mimeType) === false) {
            console.log(mimeType, 'is not supported.');
 
            // fallback to WebAudio solution
            mimeType = 'audio/wav';
            recorderType = StereoAudioRecorder;
        }
    }
}
 
var recorder = RecordRTC(mediaStream, {
    mimeType: mimeType,
    recorderType: recorderType
});

Detect Video Formats:

var mimeType = 'video/x-matroska;codecs=avc1'; // MKV
var recorderType = MediaStreamRecorder;
 
if (isMimeTypeSupported(mimeType) === false) {
    console.log(mimeType, 'is not supported.');
    mimeType = 'video/webm\;codecs=h264'; // H264
 
    if (isMimeTypeSupported(mimeType) === false) {
        console.log(mimeType, 'is not supported.');
        mimeType = 'video/webm\;codecs=vp9'; // VP9
 
        if (isMimeTypeSupported(mimeType) === false) {
            console.log(mimeType, 'is not supported.');
            mimeType = 'video/webm\;codecs=vp8'; // VP8
 
            if (isMimeTypeSupported(mimeType) === false) {
                console.log(mimeType, 'is not supported.');
                mimeType = 'video/webm'; // do NOT pass any codecs (vp8 by default)
 
                if (isMimeTypeSupported(mimeType) === false) {
                    console.log(mimeType, 'is not supported.');
 
                    // fallback to Whammy (WebP+WebM) solution
                    mimeType = 'video/webm';
                    recorderType = WhammyRecorder;
                }
            }
        }
    }
}
 
var recorder = RecordRTC(mediaStream, {
    mimeType: mimeType,
    recorderType: recorderType
});

`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);

`videoBitsPerSecond`

The chosen bitrate for the video component of the media.

var options = {
  mimeType 'video/webm', // or video/mp4
  videoBitsPerSecond: 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.

`destroy`

Destroy all internal recorders. Clear memory and ask RecordRTC to stop doing anything internally:

recorder.destroy();

Note: You can use this method anytime, anywhere; even during recording a stream.

Promises

<script src="https://cdn.WebRTC-Experiment.com/RecordRTC.js"></script>
 
<script>
// use "RecordRTCPromisesHandler" instead of "RecordRTC"
var recorder = new RecordRTCPromisesHandler(mediaStream, options);
recorder.startRecording().then(function() {
 
}).catch(function(error) {
    //
});
 
recorder.stopRecording().then(function(url) {
    var blob = recorder.blob;
 
    recorder.getDataURL().then(function(dataURL) {
        //
    }).catch(function(error) {})
}).catch(function(error) {
    //
});
</script>

Demo:

simple-demos/RecordRTCPromisesHandler.html

Credits

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

Spec & Reference

Contribute in RecordRTC.org domain

The domain https://RecordRTC.org is open-sourced here:

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

Issues/Questions?

Stackoverflow: http://stackoverflow.com/questions/tagged/recordrtc
Github: https://github.com/muaz-khan/RecordRTC/issues
Disqus: https://www.webrtc-experiment.com/RecordRTC/#ask
Email: muazkh@gmail.com

Travis Failed?

Steps to fix it (for your private projects only):

Modify package.json and search this line:

{
    "test": "./node_modules/.bin/protractor test/browserstack.config.js"
}

Replace it with (i.e. ignore all "test"):

{
    "test": "node npm-test.js"
}

Why? Reason is this file: test/browserstack.config.js

'browserstack.user': process.env.BROWSERSTACK_USERNAME,
'browserstack.key': process.env.BROWSERSTACK_KEY,

Your travis do NOT have these environment variables. That's why your travis builds fails.

More info: https://github.com/muaz-khan/RecordRTC/pull/283#issuecomment-308757116

Caution: NEVER make pull-request for modified package.json. Modify this file only for your own private projects.

Tests sponsored by

Check all tests here: https://travis-ci.org/muaz-khan/RecordRTC

Source code: https://github.com/muaz-khan/RecordRTC/tree/master/test

License

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

Keywords

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weekly downloads

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recordrtc

RecordRTC: WebRTC JavaScript Library for Audio+Video+Screen Recording

Demo: https://www.webrtc-experiment.com/RecordRTC/

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How to set video width/height?

pauseRecording

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getDataURL

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toURL

save

bufferSize

sampleRate

desiredSampRate

mimeType

isMimeTypeSupported

bitsPerSecond

audioBitsPerSecond

videoBitsPerSecond

onAudioProcessStarted

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`options`

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`onStateChanged`

`state`

`version`

Record `<canvas>`

`initRecorder`

`setRecordingDuration`

`clearRecordedData`

`recorderType`

`type`

`frameInterval`

`disableLogs`

`numberOfAudioChannels`

`pauseRecording`

`resumeRecording`

`getDataURL`

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`toURL`

`save`

`bufferSize`

`sampleRate`

`desiredSampRate`

`mimeType`

`isMimeTypeSupported`

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`audioBitsPerSecond`

`videoBitsPerSecond`

`onAudioProcessStarted`

`autoWriteToDisk`

`writeToDisk`

`getFromDisk`

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