I'm working on a project which requires the ability to stream audio from a webpage to other clients. I'm already using websocket and would like to channel the data there.
My current approach uses Media Recorder, but there is a problem with sampling which causes interrupts. It registers 1s audio and then send's it to the server which relays it to other clients. Is there a way to capture a continuous audio stream and transform it to base64?
Maybe if there is a way to create a base64 audio from MediaStream without delay it would solve the problem. What do you think?
I would like to keep using websockets, I know there is webrtc.
Have you ever done something like this, is this doable?
--> Device 1
MediaStream -> MediaRecorder -> base64 -> WebSocket -> Server --> Device ..
--> Device 18
Here a demo of the current approach... you can try it here: https://jsfiddle.net/8qhvrcbz/
var sendAudio = function(b64) {
var message = 'var audio = document.createElement(\'audio\');';
message += 'audio.src = "' + b64 + '";';
message += 'audio.play().catch(console.error);';
eval(message);
console.log(b64);
}
navigator.mediaDevices.getUserMedia({
audio: true
}).then(function(stream) {
setInterval(function() {
var chunks = [];
var recorder = new MediaRecorder(stream);
recorder.ondataavailable = function(e) {
chunks.push(e.data);
};
recorder.onstop = function(e) {
var audioBlob = new Blob(chunks);
var reader = new FileReader();
reader.readAsDataURL(audioBlob);
reader.onloadend = function() {
var b64 = reader.result
b64 = b64.replace('application/octet-stream', 'audio/mpeg');
sendAudio(b64);
}
}
recorder.start();
setTimeout(function() {
recorder.stop();
}, 1050);
}, 1000);
});
Websocket is not the best. I solved by using WebRTC instead of websocket.
The solution with websocket was obtained while recording 1050ms instead of 1000, it causes a bit of overlay but still better than hearing blanks.
Although you have solved this through WebRTC, which is the industry recommended approach, I'd like to share my answer on this.
The problem here is not websockets in general but rather the MediaRecorder API. Instead of using it one can use PCM audio capture and then submit the captured array buffers into a web worker or WASM for encoding to MP3 chunks or similar.
const context = new AudioContext();
let leftChannel = [];
let rightChannel = [];
let recordingLength = null;
let bufferSize = 512;
let sampleRate = context.sampleRate;
const audioSource = context.createMediaStreamSource(audioStream);
const scriptNode = context.createScriptProcessor(bufferSize, 1, 1);
audioSource.connect(scriptNode);
scriptNode.connect(context.destination);
scriptNode.onaudioprocess = function(e) {
// Do something with the data, e.g. convert it to WAV or MP3
};
Based on my experiments this would give you "real-time" audio. My theory with the MediaRecorder API is that it does some buffering first before emitting out anything that causes the observable delay.
Related
I'm trying to record parts of the video from a tag, save it for later use. And I found this article: Recording a media element, which described a method by first calling stream = video.captureStream(), then use new MediaRecord(stream) to get a recorder.
I've tested on some demos, the MediaRecorder works fine if stream is from user's device (such as microphone). However, when it comes to media element, my FireFox browser throws an exception: MediaRecorder.start: The MediaStream's isolation properties disallow access from MediaRecorder.
So any idea on how to deal with it?
Browser: Firefox
The page (including the js file) is stored at local.
The src attribute of <video> tag could either be a file from local storage or a url from Internet.
Code snippets:
let chunks = [];
let getCaptureStream = function () {
let stream;
const fps = 0;
if (video.captureStream) {
console.log("use captureStream");
stream = video.captureStream(fps);
} else if (video.mozCaptureStream) {
console.log("use mozCaptureStream");
stream = video.mozCaptureStream(fps);
} else {
console.error('Stream capture is not supported');
stream = null;
}
return stream;
}
video.addEventListener('play', () => {
let stream = getCaptureStream();
const mediaRecorder = new MediaRecorder(stream);
mediaRecorder.onstop = function() {
const newVideo = document.createElement('video');
newVideo.setAttribute('controls', '');
newVideo.controls = true;
const blob = new Blob(chunks);
chunks = [];
const videoURL = window.URL.createObjectURL(blob, { 'type' : 'video/mp4; codecs="avc1.42E01E, mp4a.40.2"' });
newVideo.src = videoURL;
document.body.appendChild(video);
}
mediaRecorder.ondataavailable = function(e) {
chunks.push(e.data);
}
stopButton.onclick = function() {
mediaRecorder.stop()
}
mediaRecorder.start(); // This is the line triggers exception.
});
I found the solution myself.
When I turned to Chrome, it shows that a CORS issue limits me from even playing original video. So I guess it's because the secure strategy that preventing MediaRecorder from accessing MediaStreams. Therefore, I deployed the local files to a local server with instruction on this page.
After that, the MediaRecorder started working. Hope this will help someone in need.
But still, the official document doesn't seem to mention much about isolation properties of media elements. So any idea or further explanation is welcomed.
i made this webapp to compose music, i wanted to add a feature to download the composition as .mp3/wav/whateverFileFormatPossible, i've been searching on how to do this for many times and always gave up as i couldn't find any examples on how to do it, only things i found were microphone recorders but i want to record the final audio destination of the website.
I play audio in this way:
const a_ctx = new(window.AudioContext || window.webkitAudioContext)()
function playAudio(buf){
const source = a_ctx.createBufferSource()
source.buffer = buf
source.playbackRate.value = pitchKey;
//Other code to modify the audio like adding reverb and changing volume
source.start(0)
}
where buf is the AudioBuffer.
To sum up, i want to record the whole window audio but can't come up with a way.
link to the whole website code on github
Maybe you could use the MediaStream Recording API (https://developer.mozilla.org/en-US/docs/Web/API/MediaStream_Recording_API):
The MediaStream Recording API, sometimes simply referred to as the Media Recording API or the MediaRecorder API, is closely affiliated with the Media Capture and Streams API and the WebRTC API. The MediaStream Recording API makes it possible to capture the data generated by a MediaStream or HTMLMediaElement object for analysis, processing, or saving to disk. It's also surprisingly easy to work with.
Also, you may take a look at this topic: new MediaRecorder(stream[, options]) stream can living modify?. It seems to discuss a related issue and might provide you with some insights.
The following code generates some random noise, applies some transform, plays it and creates an audio control, which allows the noise to be downloaded from the context menu via "Save audio as..." (I needed to change the extension of the saved file to .wav in order to play it.)
<html>
<head>
<script>
const context = new(window.AudioContext || window.webkitAudioContext)()
async function run()
{
var myArrayBuffer = context.createBuffer(2, context.sampleRate, context.sampleRate);
// Fill the buffer with white noise;
// just random values between -1.0 and 1.0
for (var channel = 0; channel < myArrayBuffer.numberOfChannels; channel++) {
// This gives us the actual array that contains the data
var nowBuffering = myArrayBuffer.getChannelData(channel);
for (var i = 0; i < myArrayBuffer.length; i++) {
// audio needs to be in [-1.0; 1.0]
nowBuffering[i] = Math.random() * 2 - 1;
}
}
playAudio(myArrayBuffer)
}
function playAudio(buf){
const streamNode = context.createMediaStreamDestination();
const stream = streamNode.stream;
const recorder = new MediaRecorder( stream );
const chunks = [];
recorder.ondataavailable = evt => chunks.push( evt.data );
recorder.onstop = evt => exportAudio( new Blob( chunks ) );
const source = context.createBufferSource()
source.onended = () => recorder.stop();
source.buffer = buf
source.playbackRate.value = 0.2
source.connect( streamNode );
source.connect(context.destination);
source.start(0)
recorder.start();
}
function exportAudio( blob ) {
const aud = new Audio( URL.createObjectURL( blob ) );
aud.controls = true;
document.body.prepend( aud );
}
</script>
</head>
<body onload="javascript:run()">
<input type="button" onclick="context.resume()" value="play"/>
</body>
</html>
Is this what you were looking for?
I have a stream of numbers like this
-0.00015259254737998596,-0.00009155552842799158,0.00009155552842799158,0.00021362956633198035,0.0003662221137119663,0.0003967406231879635,0.00024414807580797754,0.00012207403790398877,0.00012207403790398877,0.00012207403790398877,0.0003357036042359691,0.0003357036042359691,0.00018311105685598315,0.00003051850947599719,0,-0.00012207403790398877,0.00006103701895199438,0.00027466658528397473,0.0003967406231879635,0.0003967406231879635,0.0003967406231879635,0.0003967406231879635,0.0003967406231879635,0.0003662221137119663,0.0004882961516159551,0.0004577776421399579,0.00027466658528397473,0.00003051850947599719,-0.00027466658528397473....
Which supposedly represent an audio stream. I got them from here and I've transmitted them over the web, now I'm trying to play the actual sound and I got a snippet from here but I'm getting Uncaught (in promise) DOMException: Unable to decode audio data
I feel like I'm missing quite a lot I just expect this to work like magic and it just could not be the case..
My code
var ws = new WebSocket("ws://....");
ws.onmessage = function (event) {
playByteArray(event.data);
}
var context = new AudioContext();
function playByteArray(byteArray) {
var arrayBuffer = new ArrayBuffer(byteArray.length);
var bufferView = new Uint8Array(arrayBuffer);
for (var i = 0; i < byteArray.length; i++) {
bufferView[i] = byteArray[i];
}
context.decodeAudioData(arrayBuffer, function (buffer) {
buf = buffer;
play();
});
}
// Play the loaded file
function play() {
// Create a source node from the buffer
var source = context.createBufferSource();
source.buffer = buf;
// Connect to the final output node (the speakers)
source.connect(context.destination);
// Play immediately
source.start(0);
}
And the broadcasting part
var ws = new WebSocket("ws://.....");
window.addEventListener("audioinput", function onAudioInput(evt) {
if (ws) {
ws.send(evt.data);
}
}, false);
audioinput.start({
bufferSize: 8192
});
It doesn't look like you're dealing with compatible audio data formats. The code you linked to is for playing byte arrays, in which case your audio data should be a (much longer) string of integer numbers from 0 to 255.
What you've got is a fairly short (as audio data goes) string of floating point numbers. I can't tell what audio format that's supposed to be, but it would require a different player.
I am recording browser audio input from the microphone, and sending it via websocket to a nodeJs service that writes the stream to a .wav file.
My problem is that the first recording comes out fine, but any subsequent recordings come out sounding very slow, about half the speed and are therefore unusable.
If I refresh the browser the first recording works again, and subsequent recordings are slowed down which is why I am sure the problem is not in the nodeJs service.
My project is an Angular 5 project.
I have pasted the code I am trying below.
I am using binary.js ->
https://cdn.jsdelivr.net/binaryjs/0.2.1/binary.min.js
this.client = BinaryClient(`ws://localhost:9001`)
createStream() {
window.Stream = this.client.createStream();
window.navigator.mediaDevices.getUserMedia({ audio: true }).then(stream => {
this.success(stream);
})
}
stopRecording() {
this.recording = false;
this.win.Stream.end();
}
success(e) {
var audioContext = window.AudioContext || window.webkitAudioContext;
var context = new audioContext();
// the sample rate is in context.sampleRate
var audioInput = context.createMediaStreamSource(e);
var bufferSize = 2048;
var recorder = context.createScriptProcessor(bufferSize, 1, 1);
}
recorder.onaudioprocess = (e) => {
if (!this.recording) return;
console.log('recording');
var left = e.inputBuffer.getChannelData(0);
this.win.Stream.write(this.convertoFloat32ToInt16(left));
}
audioInput.connect(recorder)
recorder.connect(context.destination);
}
convertoFloat32ToInt16(buffer) {
var l = buffer.length;
var buf = new Int16Array(l)
while (l--) {
buf[l] = buffer[l] * 0xFFFF; //convert to 16 bit
}
return buf.buffer
}
I am stumped as to what can be going wrong so if anyone has experience using this browser tech I would appreciate any help.
Thanks.
I've had this exact problem - your problem is the sample rate you are writing your WAV file with is incorrect.
You need to pass the sample rate used by the browser and the microphone to the node.js which writes the binary WAV file.
Client side:
After a successfull navigator.mediaDevices.getUserMedia (in your case, success function), get the sampleRate variable from the AudioContext element:
var _smapleRate = context.sampleRate;
Then pass it to the node.js listener as a parameter. In my case I used:
binaryClient.createStream({ SampleRate: _smapleRate });
Server (Node.js) side:
Use the passed SampleRate to set the WAV file's sample rate. In my case this is the code:
fileWriter = new wav.FileWriter(wavPath, {
channels: 1,
sampleRate: meta.SampleRate,
bitDepth: 16
});
This will prevent broken sounds, low pitch sounds, low or fast WAV files.
Hope this helps.
I am sending RTP video data from an Android phone to my computer. The RTP streaming works, proven with ffplay.
Now I'm trying to receive and display the stream inside Chrome (63.0.3239.84, Win10, 64Bit). I receive the RTP packets with node.js and pass them to all connected users with help of socket.io.
To display the video in an HTML5 <video>-tag, I tried a similar approach as the example in the docs of the MediaSource API. This is, what i've tried to append the video data on the fly:
var socket = io();
var ms = new MediaSource();
var mimeCodec = 'video/mp4; codecs="avc1.42C01E"';
var queue = [];
var video = document.querySelector('video');
// receiving the rtp-packets from android, forwarded by nodes socket.io
socket.on('video-fragment', function(fragment) {
queue.push(fragment.slice(11)); // cutting of rtp header to get plain NAL units
});
video.src = URL.createObjectURL(ms);
var timeout = 500;
var appendChunk = function() {
if(queue.length > 0) {
timeout = 500;
var append = function() {
var sb = ms.addSourceBuffer(mimeCodec);
sb.appendBuffer(queue.shift());
sb.addEventListener('updateend', function() {
if(video.paused) {
console.log("try to start video");
video.play();
}
appendChunk();
});
ms.removeSourceBuffer(sb);
};
if(ms.readyState == "open") {
append();
} else {
ms.addEventListener('sourceopen', function() {
appendChunk();
});
}
} else {
setTimeout(appendChunk, timeout); // try again after some timeout...
timeout *= 2;
}
}
appendChunk(); // start recursive call
<video></video>
However, I don't get any errors from my video-object. If I close the MediaSource at some point with ms.endOfStream() and try to start the video after that, I get a MediaError with code 4: DEMUXER_ERROR_COULD_NOT_OPEN.
Any suggestions? Am I somehow on the right way? Or is this kind of video streaming not possible in Chrome and the only way of "live streaming" would be with using a playlist and video-file-chunks?
The browser only supports video in an mp4 (or webm) container. It does not support the rtp protocol. It must be repackaged to a format the browser supports.