I am using recording.js. The functionality is working fine but after I stop recording the red icon still appears in chrome's tab(near title). Please suggest what to do.
Sorry if it is damn easy.. :P
This is my code:
window.URL = window.URL || window.webkitURL;
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia;
var recorder;
var savedSrc = '';
var audio = document.querySelector('audio');
var onFail = function(e)
{
console.log('Rejected!', e);
};
var onSuccess = function(s)
{
var context = new AudioContext();
var mediaStreamSource = context.createMediaStreamSource(s);
recorder = new Recorder(mediaStreamSource);
recorder.record();
$('#recordText').html('Recording...');
// audio loopback
// mediaStreamSource.connect(context.destination);
};
function startRecording()
{
if (navigator.getUserMedia)
{
navigator.getUserMedia(
{
video : false,
audio : true,
toString : function()
{
return "audio";
}
}, onSuccess, onFail);
}
else
{
console.log('navigator.getUserMedia not present');
}
};
function stopRecording()
{
$('#recordText').html('Record');
recorder.stop();
recorder.exportWAV(function(s)
{
audio.src = window.URL.createObjectURL(s);
});
}
To remove red icon after using Recorder.js:
var audioStream;
var onSuccess = function(s) {
...
audioStream = s;
}
function stopRecording() {
...
audioStream.getTracks()[0].stop();
}
getTracks() returns only one element since you use only audio in your config.
I hope it will help someone.
You can end the stream directly using the stream object returned in the success handler to getUserMedia.
Example
localMediaStream.stop()
It means that your browser is holding the active instance of mic stream.
The solution given below can be a food for your thoughts.
Solution:
While assigning an audio stream ensure that you assigned reference to it to windows variable and control it (stop) from wherever you need it.
See if my code can make sense to you, its working in my case. Ensure that you reassign the stream once you stopped otherwise you ll get an exception as stop completely destroy the existing instance. (please adapt the code accordingly)
// Excerpt from my reactjs code
async function requestRecorder() {
const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
window.localStream=stream;
return new MediaRecorder(stream);
}
const stopRecording = () =>
{
//release mic resource and hence red icon is removed
window.localStream.getTracks()[0].stop();
};
It's a browser feature, not a site feature. It will be there until you close the tab, indicates that "This tab has access to or using microphone or webcam".
At the time of writing the answer there were no way to remove that icon. You may now can remove it after the recording has stopped.
Check #akaravashkin 's answer, I haven't tested it.
Related
I am pretty sure I did everything correct but when I try to play or download the file nothing plays. I am using web audio api to record audio from the microphone to a WAV format. I am using this library to create the .wav file. It seems like nothing is being encoded.
navigator.mediaDevices.getUserMedia({
audio: true,video:false
})
.then((stream) => {
var data
context = new AudioContext()
var source = context.createMediaStreamSource(stream)
var scriptNode = context.createScriptProcessor(8192, 1, 1)
source.connect(scriptNode)
scriptNode.connect(context.destination)
encoder = new WavAudioEncoder(16000,1)
scriptNode.onaudioprocess = function(e){
data = e.inputBuffer.getChannelData('0')
console.log(data)
encoder.encode(data)
}
$('#stop').click(()=>{
source.disconnect()
scriptNode.disconnect()
blob = encoder.finish()
console.log(blob)
url = window.URL.createObjectURL(blob)
// audio source
$('#player').attr('src',url)
// audio control
$("#pw")[0].load()
})
})
I figured it out! To help anyone who needs to do the same thing. It uses Web Audio API and this javascript library
navigator.mediaDevices.getUserMedia({
audio: true,video:false
})
.then((stream) => {
context = new AudioContext()
var source = context.createMediaStreamSource(stream)
var rec = new Recorder(source)
rec.record()
$('#stop').click(()=>{
rec.stop()
blob = rec.exportWAV(somefunction) // exportWAV() returns your file
})
use recordRTC for recording video and audio, I used in my project, it's working well, here is the code to record audio using recordrtc.org
startRecording(event) { // call this to start recording the Audio( or video or Both)
this.recording = true;
let mediaConstraints = {
audio: true
};
// Older browsers might not implement mediaDevices at all, so we set an empty object first
if (navigator.mediaDevices === undefined) {
navigator.mediaDevices = {};
}
// Some browsers partially implement mediaDevices. We can't just assign an object
// with getUserMedia as it would overwrite existing properties.
// Here, we will just add the getUserMedia property if it's missing.
if (navigator.mediaDevices.getUserMedia === undefined) {
navigator.mediaDevices.getUserMedia = function(constraints) {
// First get ahold of the legacy getUserMedia, if present
var getUserMedia = navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
// Some browsers just don't implement it - return a rejected promise with an error
// to keep a consistent interface
if (!getUserMedia) {
return Promise.reject(new Error('getUserMedia is not implemented in this browser'));
}
// Otherwise, wrap the call to the old navigator.getUserMedia with a Promise
return new Promise(function(resolve, reject) {
getUserMedia.call(navigator, constraints, resolve, reject);
});
}
}
navigator.mediaDevices.getUserMedia(mediaConstraints)
.then(successCallback.bind(this), errorCallback.bind(this));
}
successCallback(stream: MediaStream) {
var options = {
type: 'audio'
};
this.stream = stream;
this.recordRTC = RecordRTC(stream, options);
this.recordRTC.startRecording();
}
errorCallback(stream: MediaStream) {
console.log(stream);
}
stopRecording() { // call this to stop recording
this.recording = false;
this.converting = true;
let recordRTC = this.recordRTC;
if(!recordRTC) return;
recordRTC.stopRecording(this.processAudio.bind(this));
this.stream.getAudioTracks().forEach(track => track.stop());
}
processAudio(audioVideoWebMURL) {
let recordRTC = this.recordRTC;
var recordedBlob = recordRTC.getBlob(); // you can save the recorded media data in various formats, refer the link below.
console.log(recordedBlob)
this.recordRTC.save('audiorecording.wav');
let base64Data = '';
this.recordRTC.getDataURL((dataURL) => {
base64Data = dataURL.split('base64,')[1];
console.log(RecordRTC.getFromDisk('audio', function(dataURL,type) {
type == 'audio'
}));
console.log(dataURL);
})
}
Note that you cannot record the audio/video from the live site in Google Chrome unless your site is https enabled
I am trying to figure out how to change the microphone or webcam while you are in a videochat with someone.
I have been now trying for a few days and nothing works.
I was following this example, but it seems it is much harder to achieve the change while someone is already connected.
The issues I have: If I change the mic the sound is lost/the mic doesnt react at all. I also cannot change it back to the default.
A similar thing happens if I change the webcam. The stream hangs, the last frame is seen.
I get no error message, in fact it tells me that the changes were successful.
Changing the webcam/mic WORKS before the call is established
Here is the relevant codeblock. Everywhere I am reading just create new constraints and give the desired deviceId to the audio/video stream.:
function ChangeDevice() {
if (localStream) {
localStream.getTracks().forEach(track => {
track.stop();
});
}
var audioSource = audioInputSelect.value;
var videoSource = videoSelect.value;
console.log(videoSource);
console.log(audioSource);
const newConstraints = {
audio: {deviceId: audioSource ? {exact: audioSource} : undefined},
video: {deviceId: videoSource ? {exact: videoSource} : undefined}
};
navigator.mediaDevices.getUserMedia(newConstraints).then(gotStream).then(gotDevices).catch(handleError);
}
function gotStream(stream) {
console.log('Adding local stream.');
localStream = stream;
localVideo.srcObject = stream;
sendMessage(['got user media', room]);
if (isInitiator) {
maybeStart();
}
return navigator.mediaDevices.enumerateDevices(); // I added this
}
I think these two are the relevant functions, ChangeDevice is called when I select a new device from a dropdown. The id's are correct.
Here is the whole code I use:
pastebin.com/6JrK4jJD
Luckily replaceTrack seems to work now on all browsers, so there is no need to renegotiate.
I had to edit my gotStream function like this:
function gotStream(stream) {
// If already started
// Need this if webcam or mic changes
if (isStarted) {
var videoTrack = stream.getVideoTracks()[0];
var audioTrack = stream.getAudioTracks()[0];
var sender = pc.getSenders().find(function(s) {
return s.track.kind == videoTrack.kind;
});
var sender2 = pc.getSenders().find(function(s) {
return s.track.kind == audioTrack.kind;
});
console.log('found sender:', sender);
sender.replaceTrack(videoTrack);
sender2.replaceTrack(audioTrack);
localStream = stream;
localVideo.srcObject = stream;
} else {
console.log('Adding local stream.');
localStream = stream;
localVideo.srcObject = stream;
sendMessage(['got user media', room]);
if (isInitiator) {
maybeStart();
}
}
return navigator.mediaDevices.enumerateDevices(); // I added this
}
I have implemented video chat feature already and the code is mentioned in the snippet below. But I am trying to make ScreenSharing work using the same PeerJS.
With reference to the answer here, it says we have to get a screen sharing stream from getUserMedia instead of a webcam video stream. But exactly how do I do this in the following code:
var n = <any>navigator;
n.getUserMedia = ( n.getUserMedia || n.webkitGetUserMedia || n.mozGetUserMedia || n.msGetUserMedia );
n.mediaDevices.getUserMedia({video: true})
.then((stream) => {
this.localStream = stream;
video.src = window.URL.createObjectURL(stream);
video.play();
});
I understand what the above mentioned answer says, but how do I actually get a screen sharing stream from getUserMedia and change the above code.
Basically the code you provided saves the stream (passed into the callback function to this.localStream. If you have a variable called screenShareStream, then this is the code you need:
// Your variable to refer to the stream:
var screenShareStream
var n = <any>navigator;
n.getUserMedia = ( n.getUserMedia || n.webkitGetUserMedia || n.mozGetUserMedia || n.msGetUserMedia );
n.mediaDevices.getUserMedia({video: true})
.then((stream) => {
screenShareStream = stream;
video.src = window.URL.createObjectURL(stream);
video.play();
});
I'm working on a project and I require to send an audio stream to a Node.js server. I'm able to capture microphone sound with this function:
function micCapture(){
'use strict';
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
var constraints = {
audio: true,
video: false
};
var video = document.querySelector('video');
function successCallback(stream) {
window.stream = stream; // stream available to console
if (window.URL) {
video.src = window.webkitURL.createObjectURL(stream);
} else {
video.src = stream;
}
//Send audio stream
//server.send(stream);
}
function errorCallback(error) {
console.log('navigator.getUserMedia error: ', error);
}
navigator.getUserMedia(constraints, successCallback, errorCallback);
}
As you can see, I'm able to capture audio and play it on the website.
Now I want to send that audio stream to a Node.js server, and send it back to other clients. Like a voicechat, but I don't want to use WebRTC as I need the stream in the server. How can I achieve this? Can I use socket.io-stream to do this? In the examples I saw, they recorded the audio, and sent a file, but I need "live" audio.
I have recently done live audio upload using socket.io from browser to server. I am going to answer here in case someone else needs it.
var stream;
var socket = io();
var bufferSize = 1024 * 16;
var audioContext = new AudioContext();
// createScriptProcessor is deprecated. Let me know if anyone find alternative
var processor = audioContext.createScriptProcessor(bufferSize, 1, 1);
processor.connect(audioContext.destination);
navigator.mediaDevices.getUserMedia({ video: false, audio: true }).then(handleMicStream).catch(err => {
console.log('error from getUserMedia', err);
});
handleMicStream will run when user accepts the permission to use microphone.
function handleMicStream(streamObj) {
// keep the context in a global variable
stream = streamObj;
input = audioContext.createMediaStreamSource(stream);
input.connect(processor);
processor.onaudioprocess = e => {
microphoneProcess(e); // receives data from microphone
};
}
function microphoneProcess(e) {
const left = e.inputBuffer.getChannelData(0); // get only one audio channel
const left16 = convertFloat32ToInt16(left); // skip if you don't need this
socket.emit('micBinaryStream', left16); // send to server via web socket
}
// Converts data to BINARY16
function convertFloat32ToInt16(buffer) {
let l = buffer.length;
const buf = new Int16Array(l / 3);
while (l--) {
if (l % 3 === 0) {
buf[l / 3] = buffer[l] * 0xFFFF;
}
}
return buf.buffer;
}
Have your socket.io server listen to micBinaryStream and you should get the data. I needed the data as a BINARY16 format for google api if you do not need this you can skip the function call to convertFloat32ToInt16().
Important
When you need to stop listening you MUST disconnect the the processor and end the stream. Run the function closeAll() below.
function closeAll() {
const tracks = stream ? stream.getTracks() : null;
const track = tracks ? tracks[0] : null;
if (track) {
track.stop();
}
if (processor) {
if (input) {
try {
input.disconnect(processor);
} catch (error) {
console.warn('Attempt to disconnect input failed.');
}
}
processor.disconnect(audioContext.destination);
}
if (audioContext) {
audioContext.close().then(() => {
input = null;
processor = null;
audioContext = null;
});
}
}
it's an old time question, i see. I'm doing the same thing (except my server doesn't run node.js and is written in C#) and stumbled upon this.
Don't know if someone is still interested but i've elaborated a bit. The current alternative to the deprecated createScriptProcessor is the AudioWorklet interface.
From: https://webaudio.github.io/web-audio-api/#audioworklet
1.32.1. Concepts
The AudioWorklet object allows developers to supply scripts (such as JavaScript or >WebAssembly code) to process audio on the rendering thread, supporting custom >AudioNodes. This processing mechanism ensures synchronous execution of the script >code with other built-in AudioNodes in the audio graph.
You cannot implement interfaces in Javascript as far as i know but you can extend a class derived from it.
And the one we need is: https://developer.mozilla.org/en-US/docs/Web/API/AudioWorkletProcessor
So i did write a processor that just mirrors the output with the input values and displays them.
class CustomAudioProcessor extends AudioWorkletProcessor {
process (inputs, outputs, parameters) {
const input = inputs[0];
const output = output[0];
for (let channel = 0; channel < input.length; ++channel) {
for (let i = 0; i < input[channel].length; ++i) {
// Just copying all the data from input to output
output[channel][i] = input[channel][i];
// The next one will make the app crash but yeah, the values are there
// console.log(output[channel][i]);
}
}
}
}
The processor must then be placed into the audio pipeline, after the microphone and before the speakers.
function record() {
constraints = { audio: true };
navigator.mediaDevices.getUserMedia(constraints)
.then(function(stream) {
audioCtx = new AudioContext();
var source = audioCtx.createMediaStreamSource(stream);
audioCtx.audioWorklet.addModule("custom-audio-processor.js").then(() => {
customAudioProcessor = new AudioWorkletNode(audioCtx, "custom-audio-processor");
source.connect(customAudioProcessor);
customAudioProcessor.connect(audioCtx.destination);
})
audioCtx.destination.play();
Works! Good luck! :)
Is it possible to access the microphone (built-in or auxiliary) from a browser using client-side JavaScript?
Ideally, it would store the recorded audio in the browser. Thanks!
Here we capture microphone audio as a Web Audio API event loop buffer using getUserMedia() ... time domain and frequency domain snippets of each audio event loop buffer are printed (viewable in browser console just hit key F12 or ctrl+shift+i )
<html><head><meta http-equiv="Content-Type" content="text/html; charset=ISO-8859-1">
<title>capture microphone audio into buffer</title>
<script type="text/javascript">
var webaudio_tooling_obj = function () {
var audioContext = new AudioContext();
console.log("audio is starting up ...");
var BUFF_SIZE = 16384;
var audioInput = null,
microphone_stream = null,
gain_node = null,
script_processor_node = null,
script_processor_fft_node = null,
analyserNode = null;
if (!navigator.getUserMedia)
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia || navigator.msGetUserMedia;
if (navigator.getUserMedia){
navigator.getUserMedia({audio:true},
function(stream) {
start_microphone(stream);
},
function(e) {
alert('Error capturing audio.');
}
);
} else { alert('getUserMedia not supported in this browser.'); }
// ---
function show_some_data(given_typed_array, num_row_to_display, label) {
var size_buffer = given_typed_array.length;
var index = 0;
var max_index = num_row_to_display;
console.log("__________ " + label);
for (; index < max_index && index < size_buffer; index += 1) {
console.log(given_typed_array[index]);
}
}
function process_microphone_buffer(event) { // invoked by event loop
var i, N, inp, microphone_output_buffer;
microphone_output_buffer = event.inputBuffer.getChannelData(0); // just mono - 1 channel for now
// microphone_output_buffer <-- this buffer contains current gulp of data size BUFF_SIZE
show_some_data(microphone_output_buffer, 5, "from getChannelData");
}
function start_microphone(stream){
gain_node = audioContext.createGain();
gain_node.connect( audioContext.destination );
microphone_stream = audioContext.createMediaStreamSource(stream);
microphone_stream.connect(gain_node);
script_processor_node = audioContext.createScriptProcessor(BUFF_SIZE, 1, 1);
script_processor_node.onaudioprocess = process_microphone_buffer;
microphone_stream.connect(script_processor_node);
// --- enable volume control for output speakers
document.getElementById('volume').addEventListener('change', function() {
var curr_volume = this.value;
gain_node.gain.value = curr_volume;
console.log("curr_volume ", curr_volume);
});
// --- setup FFT
script_processor_fft_node = audioContext.createScriptProcessor(2048, 1, 1);
script_processor_fft_node.connect(gain_node);
analyserNode = audioContext.createAnalyser();
analyserNode.smoothingTimeConstant = 0;
analyserNode.fftSize = 2048;
microphone_stream.connect(analyserNode);
analyserNode.connect(script_processor_fft_node);
script_processor_fft_node.onaudioprocess = function() {
// get the average for the first channel
var array = new Uint8Array(analyserNode.frequencyBinCount);
analyserNode.getByteFrequencyData(array);
// draw the spectrogram
if (microphone_stream.playbackState == microphone_stream.PLAYING_STATE) {
show_some_data(array, 5, "from fft");
}
};
}
}(); // webaudio_tooling_obj = function()
</script>
</head>
<body>
<p>Volume</p>
<input id="volume" type="range" min="0" max="1" step="0.1" value="0.5"/>
</body>
</html>
Since this code exposes microphone data as a buffer you could add ability to stream using websockets or simply aggregate each event loop buffer into a monster buffer then download the monster to a file
Notice the call to
var audioContext = new AudioContext();
which indicates its using the Web Audio API which is baked into all modern browsers (including mobile browsers) to provide an extremely powerful audio platform of which tapping into the mic is but a tiny fragment ... NOTE the CPU usage jumps up due to this demo writing each event loop buffer into browser console log which is for testing only so actual use is far less resource intensive even when you mod this to stream audio to elsewhere
Links to some Web Audio API documentation
Basic concepts behind Web Audio API
SO wiki on Web Audio API
nice Web Audio API demos ... some with github links
Yes you can.
Using the getUserMedia() API, you can capture raw audio input from your microphone.
In a secure context, to query the devices.
getUserMedia() is a powerful feature which can only be used in secure
contexts; in insecure contexts, navigator.mediaDevices is undefined,
preventing access to getUserMedia(). A secure context is, in short, a
page loaded using HTTPS or the file:/// URL scheme, or a page loaded
from localhost.
async function getMedia(constraints) {
let stream = null;
try {
stream = await navigator.mediaDevices.getUserMedia(constraints);
console.log(stream)
} catch(err) {
document.write(err)
}
}
getMedia({ audio: true, video: true })
https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
This is a simple way:
//event:
const micButtonClicked = () => {
//check the access:
isMicrophoneAllowed(isAllowed => {
if(isAllowed)
record();
else
navigator.mediaDevices.getUserMedia({audio: true})
.then(stream => record())
.catch(err => alert('need permission to use microphone'));
});
}
//isMicrophoneAllowed:
const isMicrophoneAllowed = callback => {
navigator.permissions.query({name: 'microphone'})
.then(permissionStatus => Strings.runCB(callback, permissionStatus.state === 'granted'));
}
//record:
const record = () => {
// start recording...
}