Trying to follow the example here, which is basically a c&p of this
Think I got most of the parts down, except all the node.connect()'s
From what I understand, this sequence of code is needed to provide the audio analyzer with an audio stream:
var source = audioCtx.createMediaStreamSource(stream);
source.connect(analyser);
analyser.connect(audioCtx.destination);
I can't seem to make sense of it as it looks rather ouroboros-y to me.
And unfortunately, I can't seem to find any documentation on .connect() so quite lost and would appreciate any clarification!
Oh and I'm loading an .mp3 via pure javascript new Audio('db.mp3').play(); and am trying to use that as the source without creating an <audio> element.
Can a mediaStream object be created from this to feed into .createMediaStreamSource(stream)?
connect simply defines the output for the filters.
In this case, your source loads the stream into the buffer and writes to the input of the next filter which is defined by the connect function. This is repeated for your analyser filter.
Think of it as pipes.
here is a sample code snippet that I have written a few years back using web audio api.
this.scriptProcessor = this.audioContext.createScriptProcessor(this.scriptProcessorBufferSize,
this.scriptProcessorInputChannels,
this.scriptProcessorOutputChannels);
this.scriptProcessor.connect(this.audioContext.destination);
this.scriptProcessor.onaudioprocess = updateMediaControl.bind(this);
//Set up the Gain Node with a default value of 1(max volume).
this.gainNode = this.audioContext.createGain();
this.gainNode.connect(this.audioContext.destination);
this.gainNode.gain.value = 1;
sewi.AudioResourceViewer.prototype.playAudio = function(){
if(this.audioBuffer){
this.source = this.audioContext.createBufferSource();
this.source.buffer = this.audioBuffer;
this.source.connect(this.gainNode);
this.source.connect(this.scriptProcessor);
this.beginTime = Date.now();
this.source.start(0, this.offset);
this.isPlaying = true;
this.controls.update({playing: this.isPlaying});
updateGraphPlaybackPosition.call(this, this.offset);
}
};
So as you can see that my source is connected to a gainNode, which is connected to a scriptProcessor. When the audio starts playing, the data is passed from the source->gainNode->destination and source->scriptProcessor->destination. flowing through the "pipes" that connects them, which is defined by connect(). When the audio data pass through the gainNode, volume can be adjusted by changing the amplitude of the audio wave. After that it is passed to the script processor so that events can be attached and triggered while the audio is being processed.
Related
I already looked at this question -
Concatenate parts of two or more webm video blobs
And tried the sample code here - https://developer.mozilla.org/en-US/docs/Web/API/MediaSource -- (without modifications) in hopes of transforming the blobs into arraybuffers and appending those to a sourcebuffer for the MediaSource WebAPI, but even the sample code wasn't working on my chrome browser for which it is said to be compatible.
The crux of my problem is that I can't combine multiple blob webm clips into one without incorrect playback after the first time it plays. To go straight to the problem please scroll to the line after the first two chunks of code, for background continue reading.
I am designing a web application that allows a presenter to record scenes of him/herself explaining charts and videos.
I am using the MediaRecorder WebAPI to record video on chrome/firefox. (Side question - is there any other way (besides flash) that I can record video/audio via webcam & mic? Because MediaRecorder is not supported on not Chrome/Firefox user agents).
navigator.mediaDevices.getUserMedia(constraints)
.then(gotMedia)
.catch(e => { console.error('getUserMedia() failed: ' + e); });
function gotMedia(stream) {
recording = true;
theStream = stream;
vid.src = URL.createObjectURL(theStream);
try {
recorder = new MediaRecorder(stream);
} catch (e) {
console.error('Exception while creating MediaRecorder: ' + e);
return;
}
theRecorder = recorder;
recorder.ondataavailable =
(event) => {
tempScene.push(event.data);
};
theRecorder.start(100);
}
function finishRecording() {
recording = false;
theRecorder.stop();
theStream.getTracks().forEach(track => { track.stop(); });
while(tempScene[0].size != 1) {
tempScene.splice(0,1);
}
console.log(tempScene);
scenes.push(tempScene);
tempScene = [];
}
The function finishRecording gets called and a scene (an array of blobs of mimetype 'video/webm') gets saved to the scenes array. After it gets saved. The user can then record and save more scenes via this process. He can then view a certain scene using this following chunk of code.
function showScene(sceneNum) {
var sceneBlob = new Blob(scenes[sceneNum], {type: 'video/webm; codecs=vorbis,vp8'});
vid.src = URL.createObjectURL(sceneBlob);
vid.play();
}
In the above code what happens is the blob array for the scene gets turning into one big blob for which a url is created and pointed to by the video's src attribute, so -
[blob, blob, blob] => sceneBlob (an object, not array)
Up until this point everything works fine and dandy. Here is where the issue starts
I try to merge all the scenes into one by combining the blob arrays for each scene into one long blob array. The point of this functionality is so that the user can order the scenes however he/she deems fit and so he can choose not to include a scene. So they aren't necessarily in the same order as they were recorded in, so -
scene 1: [blob-1, blob-1] scene 2: [blob-2, blob-2]
final: [blob-2, blob-2, blob-1, blob-1]
and then I make a blob of the final blob array, so -
final: [blob, blob, blob, blob] => finalBlob
The code is below for merging the scene blob arrays
function mergeScenes() {
scenes[scenes.length] = [];
for(var i = 0; i < scenes.length - 1; i++) {
scenes[scenes.length - 1] = scenes[scenes.length - 1].concat(scenes[i]);
}
mergedScenes = scenes[scenes.length - 1];
console.log(scenes[scenes.length - 1]);
}
This final scene can be viewed by using the showScene function in the second small chunk of code because it is appended as the last scene in the scenes array. When the video is played with the showScene function it plays all the scenes all the way through. However, if I press play on the video after it plays through the first time, it only plays the last scene.
Also, if I download and play the video through my browser, the first time around it plays correctly - the subsequent times, I see the same error.
What am I doing wrong? How can I merge the files into one video containing all the scenes? Thank you very much for your time in reading this and helping me, and please let me know if I need to clarify anything.
I am using a element to display the scenes
The file's headers (metadata) should only be appended to the first chunk of data you've got.
You can't make an new video file by just pasting one after the other, they've got a structure.
So how to workaround this ?
If I understood correctly your problem, what you need is to be able to merge all the recorded videos, just like if it were only paused.
Well this can be achieved, thanks to the MediaRecorder.pause() method.
You can keep the stream open, and simply pause the MediaRecorder. At each pause event, you'll be able to generate a new video containing all the frames from the beginning of the recording, until this event.
Here is an external demo because stacksnippets don't works well with gUM...
And if ever you needed to also have shorter videos from between each resume and pause events, you could simply create new MediaRecorders for these smaller parts, while keeping the big one running.
I've been trying to create polyphonic WAV playback with node.js on raspberry pi 3 running latest raspbian:
shelling out to aplay/mpg123/some other program - allows me to only play single sound at once
I tried combination of https://github.com/sebpiq/node-web-audio-api and https://github.com/TooTallNate/node-speaker (sample code below) but audio quality is very low, with a lot of distortions
Is there anything I'm missing here? I know I could easily do it in another programming language (I was able to write C++ code with SDL, and Python with pygame), but the question is if it's possible with node.js :)
Here's my current web-audio-api + node-speaker code:
var AudioContext = require('web-audio-api').AudioContext;
var Speaker = require('speaker');
var fs = require('fs');
var track1 = './tracks/1.wav';
var track2 = './tracks/1.wav';
var context = new AudioContext();
context.outStream = new Speaker({
channels: context.format.numberOfChannels,
bitDepth: context.format.bitDepth,
sampleRate: context.format.sampleRate
});
function play(audioBuffer) {
if (!audioBuffer) { return; }
var bufferSource = context.createBufferSource();
bufferSource.connect(context.destination);
bufferSource.buffer = audioBuffer;
bufferSource.loop = false;
bufferSource.start(0);
}
var audioData1 = fs.readFileSync(track1);
var audioData2 = fs.readFileSync(track2);
var audioBuffer1, audioBuffer2;
context.decodeAudioData(audioData1, function(audioBuffer) {
audioBuffer1 = audioBuffer;
if (audioBuffer1 && audioBuffer2) { playBoth(); }
});
context.decodeAudioData(audioData2, function(audioBuffer) {
audioBuffer2 = audioBuffer;
if (audioBuffer1 && audioBuffer2) { playBoth(); }
});
function playBoth() {
console.log('playing...');
play(audioBuffer1);
play(audioBuffer2);
}
audio quality is very low, with a lot of distortions
According to the WebAudio spec (https://webaudio.github.io/web-audio-api/#SummingJunction):
No clipping is applied at the inputs or outputs of the AudioNode to allow a maximum of dynamic range within the audio graph.
Now if you're playing two audio streams, it's possible that summing them results in a value that's beyond the acceptable range, which sounds like - distortions.
Try lowering the volume of each audio stream by first piping them through a GainNode as so:
function play(audioBuffer) {
if (!audioBuffer) { return; }
var bufferSource = context.createBufferSource();
var gainNode = context.createGain();
gainNode.gain.value = 0.5 // for instance, find a good value
bufferSource.connect(gainNode);
gainNode.connect(context.destination);
bufferSource.buffer = audioBuffer;
bufferSource.loop = false;
bufferSource.start(0);
}
Alternatively, you could use a DynamicsCompressorNode, but manually setting the gain gives you more control over the output.
This isn't exactly answer-worthy but I can't post comments at the moment ><
I had a similar problem with an app made using js audio api and the, rather easy fix, was lowering the quality of the audio and changing the format.
In your case what I could think of is setting the bit depth&sampling frequency as low as possible without affecting the listener's experience (e.g. 44.1kHz and 16 bit depth).
You might also try changing the format, wav, in theory, should be quite good at the job of not being CPU intensive, however, there are other uncompressed formats (e.g. .aiff)
You may try using multiple cores of the pi:
https://nodejs.org/api/cluster.html
Although this may prove a bit complicated, if you are doing the audio-streaming in parallel with other unrelated processes, you could try moving the audio on a separate CPU.
An (easy) thing you could try would be running node with more RAM, although, in your case, I doubt that I possible.
The biggest problem, however, might be the code, sadly enough I am not experienced with the modules you are using and as such can give to real advice on that (hence, why I said this is not answer worthy :p)
when you create Speaker instant, set parameter like this
channels = 1 // you can try with 1 or 2 and get the best quantity
bitDepth = 16
sampleRate = 48000 // normally 44100 for speaking and higher for music playing
You can spawn from node 2 aplay processes each playing one file. Use detached: true to allow node to continue running.
Is it possible to use the Analyser node in the offlineAudioContext to do frequency analysis?
I found out that ScriptProcessor 's onaudioprocess event still fires in the offlineAudioContext and this was the only event source I could use to call getByteFrequencyData of the Analyser Node. As below:
var offline = new offlineAudioContext(1, buffer.length, 44100);
var bufferSource = offline.createBufferSource();
bufferSource.buffer = buffer;
var analyser = offline.createAnalyser();
var scp = offline.createScriptProcessor(256, 0, 1);
bufferSource.connect(analyser);
scp.connect(offline.destination); // this is necessary for the script processor to start
var freqData = new Uint8Array(analyser.frequencyBinCount);
scp.onaudioprocess = function(){
analyser.getByteFrequencyData(freqData);
console.log(freqData);
// freqData is always the same.
};
bufferSource.start(0);
offline.startRendering();
The problem here is that freqData array is always the same and never changes. Seem like as if it is only analysing one section of the buffer.
Am I doing anything wrong here? Or the Analyser is not intended to be used in the offlineContext.
And here is the fiddle with the same code.
An alternative is to use the suspend and resume methods available for an OfflineAudioContext. After suspending, you can use your analyser node to get the desired time and/or frequency domain data. Since the context is stopped, this works just fine. If you're going to do this several times, you'll need to schedule a stop for the next time before resuming.
Unfortunately, AFAIK, only Chrome has implemented suspend for an OfflineAudioContext
The analyser isn't really intended to be used in the offlineContext; in fact, it was originally named "RealtimeAnalyser". But even more importantly, right now you won't get consistent functionality from script processors in offline contexts, either.
With the Web Audio API, I want to save audio in a buffer for later use. I've found some examples of saving audio to disk, but I only want to store it in memory. I tried connecting the output of the last AudioNode in the chain to an AudioBuffer, but it seems AudioBuffer doesn't have a method for accepting inputs.
var contextClass = (window.AudioContext || window.webkitAudioContext);
// Output compressor
var compressor = context.createDynamicsCompressor();
var compressor.connect(context.destination);
var music = context.createBufferSource();
// Load some content into music with XMLHttpRequest...
music.connect(compressor);
music.start(0);
// Set up recording buffer
var recordBuffer = context.createBuffer(2, 10000, 44100);
compressor.connect(recordBuffer);
// Failed to execute 'connect' on 'AudioNode': No function was found that matched the signature provided.
Is there something I can use instead of AudioBuffer to achieve this? Is there a way to do this without saving files to disk?
Well, turns out Recorder.js does exactly what I wanted. I thought it was only for exporting to disk, but when I looked closer I realized it can save to buffers too. Hooray!
I'm creating an audio visualizer with webgl, and have been integrating soundcloud tracks into it. I want to no be able to switch tracks, but I can either get my visualizer to work and the audio to break, or I can get the audio to work and the visualizer to break.
The two ways that I've been able to make it work are
Audio working
delete audio element
append new audio element to body
trigger play
Visualizer working
stop audio
change source
trigger play
When I have the visualizer working, the audio is totally messed up. The buffers just sound wrong, and the audio has artifacts in it (noise, beeps and bloops).
When I have the audio working, when I call analyser.getByteFrequencyData, I get an array of 0's. I presume this is because the analyser is not hooked up correctly.
The code for the audio working looks like
$('#music').trigger("pause");
currentTrackNum = currentTrackNum + 1;
var tracks = $("#tracks").data("tracks")
var currentTrack = tracks[parseInt(currentTrackNum)%tracks.length];
// Begin audio switching
analyser.disconnect();
$('#music').remove();
$('body').append('<audio id="music" preload="auto" src="'+ currentTrack["download"].toString() + '?client_id=4c6187aeda01c8ad86e556555621074f"></audio>');
startWebAudio(),
(I don't think I need the pause call. Do I?)
when I want the visualizer to work, I use this code
currentTrackNum = currentTrackNum + 1;
var tracks = $("#tracks").data("tracks")
var currentTrack = tracks[parseInt(currentTrackNum)%tracks.length];
// Begin audio switching
$("#music").attr("src", currentTrack["download"].toString() + "?client_id=4c6187aeda01c8ad86e556555621074f");
$("#songTitle").text(currentTrack["title"]);
$('#music').trigger("play");
The startWebAudio function looks like this.
function startWebAudio() {
// Get our <audio> element
var audio = document.getElementById('music');
// Create a new audio context (that allows us to do all the Web Audio stuff)
var audioContext = new webkitAudioContext();
// Create a new analyser
analyser = audioContext.createAnalyser();
// Create a new audio source from the <audio> element
var source = audioContext.createMediaElementSource(audio);
// Connect up the output from the audio source to the input of the analyser
source.connect(analyser);
// Connect up the audio output of the analyser to the audioContext destination i.e. the speakers (The analyser takes the output of the <audio> element and swallows it. If we want to hear the sound of the <audio> element then we need to re-route the analyser's output to the speakers)
analyser.connect(audioContext.destination);
// Get the <audio> element started
audio.play();
var freqByteData = new Uint8Array(analyser.frequencyBinCount);
}
My suspicion is that the analyzer isn't hooked up correctly, but I can't figure out what to look at to figure it out. I have looked at the frequencyByteData output, and that seems to be indicative of something not being hooked up right. The analyser variable is global. If you would like more reference to the code, here's where it is on github
You can only create a single AudioContext per window. You should also be disconnecting the MediaElementSource when you're finished using it.
Here's an example that I used to answer a similar question: http://jsbin.com/acolet/1/