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I developed the recording using the JavaScript web audio API, but the sound quality is poor

I have created an application that sings along in the app with the web audio API of JavaScript. This worked perfectly on iOS safari and Chrome, but the sound quality was poor on Android Chrome. To solve this, I tried changing the audio deviceId, but it still didn't work. Does someone have information that might help?
Doubt: After recording, I pass the file to the server and play it on another page. I am wondering if this is causing the problem.
This is my code
function captureUserMedia(mediaConstraints) {
navigator.mediaDevices.getUserMedia(mediaConstraints).then(onMediaSuccess)["catch"]();
}
function record() {
if (getParameterByName("startSec").length !== 0) {
masterSound.currentTime = getParameterByName("startSec");
}
masterSound.play();
if (document.querySelectorAll(".record")[0].getAttribute("status") == "off") {
document.querySelectorAll(".record")[0].setAttribute("status", "on");
document.querySelectorAll(".record")[0].classList.add("stoped");
var mediaConstraints;
const devices = navigator.mediaDevices.enumerateDevices()
devices.then((value) => {
// mediaConstraints = {
// audio: {
// deviceId: {
// exact: value[0].deviceId
// }
// },
// video: false
// };
mediaConstraints = {
audio: true,
video: false,
};
captureUserMedia(mediaConstraints, onMediaSuccess);
});
} else {
document.querySelectorAll(".record")[0].setAttribute("status", "off");
document.querySelectorAll(".record")[0].classList.remove("stoped");
mediaRecorder.stream.stop();
masterSound.pause();
}
}
function onMediaSuccess(stream) {
var audio = document.createElement('audio');
audio.controls = true;
audio.files = true;
audio.muted = true;
audio.srcObject = stream;
audio.play();
var audiosContainer = document.querySelectorAll(".audio_wrapper")[0];
audiosContainer.appendChild(audio);
audiosContainer.appendChild(document.createElement('hr'));
mediaRecorder = new MediaStreamRecorder(stream);
mediaRecorder.mimeType = 'audio/wav';
mediaRecorder.stream = stream;
mediaRecorder.recorderType = MediaRecorderWrapper;
mediaRecorder.audioChannels = 1;
mediaRecorder.start();
mediaRecorder.ondataavailable = function (blob) {
audioFile = blob;
var blobURL = URL.createObjectURL(blob);
document.querySelectorAll(".append_audio")[0].setAttribute("src", blobURL);
function blobToFile(theBlob, fileName) {
theBlob.lastModifiedDate = new Date();
theBlob.name = fileName;
return theBlob;
}
submit();
function submit() {
var audioTest = new Audio(URL.createObjectURL(blob));
audioTest.play();
}
};
}

When trying to build high-quality audio with getDisplayMedia, in the past I've passed in MediaStreamConstraints that remove some of the default processing on the input track:
stream = await navigator.mediaDevices.getDisplayMedia(
{
video: true,
audio:
{
channels: 2,
autoGainControl: false,
echoCancellation: false,
noiseSuppression: false
}
}
);
I'm still learning WebRTC myself, so I'm not sure if these same properties can be passed when using getUserMedia and MediaConstraints, but I thought I'd share in case helpful. It sounds like this might also be about available devices. Good luck!

Had a similar issue where we were getting complaints about very low sound/gain - barely hearable - with our HTML/JS recording client when running on Chrome on some Android devices.
Ended up buying an older Samsung phone (Galaxy A8) to easily replicate the issue.
The culprit was echoCancellation being set to false. With it disabled, we had a very low volume on the recorded audio. The solution was to set echoCancellation as true.
We ended up removing the constraint altogether and relied on each browser's defaults (echoCancellation is enabled by default on Chrome, Safari, Firefox).
Worth mentioning that autoGainControl and noiseSuppression inherit the value of echoCancellation, more exactly, if you only set audio: {echoCancellation: true} the other 2 constraints will also be set as true.

RTCmulticonnection use existing getUserMedia stream

I prepared a webcam streaming app and therefore previewed a users webcam in a video tag using "navigator.mediaDevices.getUserMedia".
After the user hits a button and "RTCmulticonnection" opens a room, I want to add the preview webcam / video to be used as webcam-stream.
I just don't get it working.
The part Im going to show you starts after the socket.io Connection is established - the "connection.open" method also returns that the room has been created and the connection is established. So Signaling works - I just don't know how to simply attach my Video to the (RTCmulsiconnection) connection Object.
I have tried to use the "attachStreams" method, but remained unsuccessfully.
//Get Users Webcam
navigator.mediaDevices.getUserMedia({
audio: true,
video: true
}).then(function(stream) {
$("#mediaPreview").show();
let preview = document.getElementById('mediaPreviewVideo');
preview.srcObject = stream;
preview.volume = 0;
preview.play();
});
//Start stream
$("#btnMediaStreamStart").on('click', function(e) {
//Prevent Default & Hide
e.preventDefault();
//Add Classes
$("#mediaPreviewVideo").addClass('webcam-online');
$("#mediaStreamStatus").show().addClass('bg-success').html('<small><strong>Live</strong></small>');
connection.socketURL = 'localhost:9001/';
let connectionRoom = '123';
connection.open(connectionRoom, function(e) {
if (e === true) {
connection.attachStreams.forEach(function(localStream) {
//How to attach the cam stream? Is my previously created video even part of the "streams"?
});
connection.mediaConstraints = {
audio: {
deviceId: selectAudio
},
video: {
deviceId: selectVideo
}
};
}
});

Try using
connection.addStream(stream);
where stream is your already captured stream - perhaps via getUserMedia()

getUserMedia block reload page after user change

Is it possible to detect that the user has 'enabled' the microphone/video for the webpage without having to get the user to refresh the page manually after an access denied & manual user enable?
basically it is disabled, when the user enables it I want to refresh the page automatically.
navigator.getUserMedia (
// constraints
{
video: true,
audio: true
},
// successCallback
function(localMediaStream) {
var video = document.querySelector('video');
video.src = window.URL.createObjectURL(localMediaStream);
video.onloadedmetadata = function(e) {
// Do something with the video here.
};
},
// errorCallback
function(err) {
if(err === PERMISSION_DENIED) {
// Explain why you need permission and how to update the permission setting
}
}
);

WebRTC works in Chrome but not Firefox

I read several other questions on a related issue, but none answered my question. I have an odd issue where I am able to use WebRTC to audio chat from chrome to firefox but not firefox to chrome.
Basically, when a user wishes to audio chat, he/she clicks a button #audioChatBtn, which uses getUserMedia() to setup a stream. The thing is, clicking #audioChatBtn from Firefox doesn't fire the onaddstream callback on Chrome, but clicking the button from Chrome fires onaddstream on Firefox. Thus, I can audio chat from Chrome to Firefox but not the other way around. I have been trying to figure this out for several hours, but I'm hoping maybe someone here has an answer.
Relevant source:
var configuration = {
'iceServers': [
{ url: 'stun:stun.l.google.com:19302' },
{ url: 'stun:stun1.l.google.com:19302' },
{ url: 'stun:stun2.l.google.com:19302' },
{ url: 'stun:stun3.l.google.com:19302' },
{ url: 'stun:stun4.l.google.com:19302' }
]
};
var pc = RTCPeerConnection(configuration);
var myStream = null;
var currentAudioIndex = 0; // Number of created channels
var myAudioEnabled = false;
// send any ice candidates to the other peer
pc.onicecandidate = function (evt) {
if (evt.candidate)
$(document).trigger("persistState", { mode: 'rtc', 'candidate': evt.candidate });
};
// let the 'negotiationneeded' event trigger offer generation
pc.onnegotiationneeded = function () {
pc.createOffer(localDescCreated, logError);
}
// once remote stream arrives, play it in the audio element
pc.onaddstream = function (evt) {
console.log('creating and binding audio');
var idx = (currentAudioIndex++);
var audioElement = $('#audio' + idx);
if (audioElement.length == 0) {
var audio = $('<audio id="audio' + idx + '" autoplay>');
$('body').append(audio);
audioElement = $('#audio' + idx);
}
var audioObject = audioElement[0];
attachMediaStream(audioObject, evt.stream);
};
function localDescCreated(desc) {
pc.setLocalDescription(desc, function () {
$(document).trigger("persistState", { mode: 'rtc', 'sdp': pc.localDescription });
}, logError);
}
function logError(e) {
bootbox.alert("Audio chat could not be started.");
}
function hasGetUserMedia() {
return !!(navigator.getUserMedia || navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia || navigator.msGetUserMedia);
}
server.onPersist = function(msg) {
if (msg.mode == "rtc") {
if (msg.sdp)
pc.setRemoteDescription(new RTCSessionDescription(msg.sdp), function () {
// if we received an offer, we need to answer
if (pc.remoteDescription.type == 'offer')
pc.createAnswer(localDescCreated, logError);
}, logError);
else
pc.addIceCandidate(new RTCIceCandidate(msg.candidate));
}
}
// On click, start audio chat from this user.
$('#audioChatBtn').click(function() {
if (!hasGetUserMedia()) {
bootbox.alert('Audio conferencing is not supported by your browser. (Currently only supported by Chrome, Firefox, and Opera web browsers.)');
return;
}
if (myAudioEnabled) {
myStream.stop();
displayAlert('Streaming closed', 'Audio chat is off');
$('#audioChatBtn').removeClass('btn-success').addClass('btn-primary');
} else {
getUserMedia({ video: false, audio: true }, function (localMediaStream) {
myStream = localMediaStream;
pc.addStream(localMediaStream);
displayAlert('Streaming...', 'Audio chat is enabled');
$('#audioChatBtn').removeClass('btn-primary').addClass('btn-success');
}, logError);
}
myAudioEnabled = !myAudioEnabled;
});
What I've tried
Tried using 'optional': [{ 'DtlsSrtpKeyAgreement': 'true' }] in the configuration after reading this question
Tried creating a new RTCPeerConnection() each request
Tried using native browser functions instead of adapter.js.
Explored Web Audio API instead of getUserMedia()

Firefox does not currently support onnegotiationneeded, because we currently don't support re-negotiation of an existing connection. All addStream/addTrack and a single createDataChannel (if you want to use them) need to be done before createOffer() or createAnswer. You can createDataChannel() after you connect, if you created on before createOffer.
Adding a stream after they're connected won't work.
An (annoying) alternative is to create a new set of PeerConnections to replace the old ones (using a DataChannel in the old pair as a signaling channel for lower latency)
Resolving this is high on our priority list, but will take a few more releases.

After a lot of debugging, I came to realize that the bug has nothing to do with my code but has to do with Firefox's implementation of WebRTC. Firefox doesn't trigger the onnegotiationneeded callback, so I have to do it hackily using a timeout (and hope that the stream information has been relayed to the remote client before the function fires). Obviously, this is a firefox bug and I will report it, hoping that they fix the bug in the next build.
getUserMedia({ video: false, audio: true }, function (localMediaStream) {
myStream = localMediaStream;
pc.addStream(localMediaStream);
displayAlert('Streaming...', 'Audio chat is enabled');
$('#audioChatBtn').removeClass('btn-primary').addClass('btn-success');
// Need this for Firefox
if (webrtcDetectedBrowser == 'firefox')
setTimeout(pc.onnegotiationneeded, 5000);
}, logError);

"hello world" on webcam capturing in HTML5 fails to work

after a couple of hours of struggling here I am. I have the following code, which apparently should just start my webcam and prompt the stream on the webpage:
<!doctype html>
<html>
<head>
<title>HTML5 Webcam Test</title>
</head>
<body>
<video id="sourcevid" autoplay>Put your fallback message here.</video>
<div id="errorMessage"></div>
<script>
video = document.getElementById('sourcevid');
navigator.getUserMedia = navigator.webkitGetUserMedia || navigator.getUserMedia;
window.URL = window.URL || window.webkitURL;
function gotStream(stream) {
if (window.URL) {
video.src = window.URL.createObjectURL(stream);
} else {
video.src = stream; // Opera.
}
video.onerror = function(e) {
stream.stop();
};
stream.onended = noStream;
}
function noStream(e) {
var msg = 'No camera available.';
if (e.code == 1) {
msg = 'User denied access to use camera.';
}
document.getElementById('errorMessage').textContent = msg;
}
navigator.webkitGetUserMedia({video: true}, gotStream, noStream);
</script>
</body>
</html>
No errors in the console, but no webcam stream either. Just the "User denied access to use camera.".
I tried another example, too long to show, but again apparently as soon as I try to run the page the stream falls into the .onended function:
function gotStream(stream) {
video.src = URL.createObjectURL(stream);
video.onerror = function () {
stream.stop();
};
stream.onended = noStream;
[...]
Where noStream is a simple function that prints something:
function noStream() {
document.getElementById('errorMessage').textContent = 'No camera available.';
}
So basically when I'm running the second example I'm shown the "No camera available" on the webpage.
I'm running on Chrome Version 22.0.1229.94, I saw somewhere that I needed to enable some flags, but I couldn't find them in my chrome://flags; the flags' name were Enable MediaStream and Enable PeerConnection, but in my version I only have the second one, which I enabled.
Any thoughts? Is the API I'm using old by any means? Can somebody point me to some working example?
Thanks

According to http://www.webrtc.org/running-the-demos the getUserMedia API is available on stable version as of Chrome 21 without the need to use any flag.
I think the error happens because you are trying to instantiate the stream without to define the url stream properly. Consider that you need to access the url stream differently in Chrome and Opera.
I would create the structure of your code as something like below:
function gotStream(stream) {
if (window.URL) {
video.src = window.URL.createObjectURL(stream) || stream;
video.play();
} else {
video.src = stream || stream; // Opera.
video.play();
}
video.onerror = function(e) {
stream.stop();
};
stream.onended = noStream;
}
function noStream(e) {
var msg = 'No camera available.';
if (e.code == 1) {
msg = 'User denied access to use camera.';
}
document.getElementById('errorMessage').textContent = msg;
}
var options = {video: true, toString: function(){return 'video';}};
navigator.getUserMedia(options, gotStream, noStream);
EDIT:
You need to replace the source video element with the media stream. Edited the code above.
video = document.getElementById('sourcevid');
I recommend for reading these two articles:
http://www.html5rocks.com/en/tutorials/getusermedia/intro/
http://dev.opera.com/articles/view/playing-with-html5-video-and-getusermedia-support/