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Web MediaRecorder API : concatenating video chunks end-pads with blank video

I have working code (below) that concatenates the first and last chunk of video recorded from the browser.
Chunks are collected every three seconds, and the resultant downloaded video does indeed play the first and last chunk in sequence.
However, the total length of the video is the length of all collected chunks, rather than just the sum of the first and last. The "extra" video is just blank.
More to follow, after the code:
let mediaRecorder = new MediaRecorder()
let chunks = []
mediaRecorder.start(3000) // triggers ondataavailable every 3 seconds
mediaRecorder.ondataavailable = (ev) => {
chunks.push(ev.data);
if (chunks.length > 2) chunks.splice(1,1) // keeps only the first and last chunk
}
let blob = new Blob(chunks, {'type': 'video/mp4;'}
let videoURL = window.URL.createObjectURL(blob)
let downloadLink = document.getElementById('downloadLink');
downloadLink.href = videoURL;
mediaRecorder.stop();
downloadLink.click()
Maybe ev.timeStamp is being used to determine the video's length, rather than just the sum of the duration of the chunks? That would make more sense if the last chunk were playing back in the last 3 seconds, with blank video in the middle. But no -- it plays [beginning chunk, end chunk, blank space].
Is there a way to prevent the blank video that follows/pads the concatenated beginning/end?
--
Edit: I've also tried slicing off the end of the blob, blob.slice(0, -10,000), but it doesn't trim the blank space.

tl;dr You're skipping some MediaRecorder chunks. You Can't Do That™.
If this is on Chromium-based or Firefox browsers, MediaRecorder is generating media in a webm / Matroska boxing format. There's MKVToolNix to look at Matroska files; you can do it with a hex dump but that's an over-the-top pain in the xxx neck.
The a/v material comes in Clusters. Clusters have absolute timestamps (containing times since the beginning of the recording, not since the previous cluster) on them. And, each Cluster contains a bunch of SimpleBlocks holding the media. Although this isn't spelled out in any specs, Chromium emits a video interframe (a/k/a keyframe a/k/a Instantaneous Decoder Refesh frame) into the first video SimpleBlock in the beginning of each Cluster.
So, if you only save the last chunk, it contains a Cluster with timestamp far in the future from the beginning of the recording. The player waits til that time arrives, then plays it.
It's a small miracle what you're doing works as well as it does. For stored MediaRecorder output to work predictably you can't skip any chunks, nor can you take them out of order. For one thing, it's a pure implementation detail that the chunks break along Cluster boundaries: nothing in the spec guarantees that. I've seen MediaRecorder break chunks in other places.
Cutting out a hunk of video timeline from a Matroska / webm stream made by MediaRecorder takes a whole mess of sophisticated parsing and reassembly of the Matroska boxing. And unless you always cut right before interframes, it takes decoding and recoding the video stream too.
There's the npm ebml package to parse the Matroska data stream.

How to generate negative audio waves from input sound in javascript?

I found a javascript that captures the current microphone input just to send it out again. You can see it here:
https://codepen.io/MyXoToD/pen/bdb1b834b15aaa4b4fcc8c7b50c23a6f?editors=1010 (only works with https).
I was wondering how I can generate the "negative" waves of the sound captured. Just like "noise cancellation" works. So the script detects the current noise around me and whenever there is a wave going up I want to generate a wave that is going down to cancel each other out. Is there a way to do this in javascript?
So this outputs the current sound recored:
var input = audio_context.createMediaStreamSource(stream);
volume = audio_context.createGain();
volume.gain.value = 0.8;
input.connect(volume);
volume.connect(audio_context.destination);
The question is how to produce the negative of this sound and output this instead.

You're already streaming your signal through a Web Audio chain. What you need to do is chain an AudioWorker node between the input and output nodes (after or before the gain node, doesn't matter). The script for the worker would be a very simple sign inversion.
var negationWorker = audio_context.createAudioWorker( "negate.js" )
var negationNode = negationWorker.createNode()
input.connect(negationNode)
negationNode.connect(volume)

Get Final Output Frequency of Chained Oscillators

I've set up a web page with a theremin and I'm trying to change the color of a web page element based on the frequency of the note being played. The way I'm generating sound right now looks like this:
osc1 = page.audioCX.createOscillator();
pos = getMousePos(page.canvas, ev);
osc1.frequency.value = pos.x;
gain = page.audioCX.createGain();
gain.gain.value = 60;
osc2 = page.audioCX.createOscillator();
osc2.frequency.value = 1;
osc2.connect(gain);
gain.connect(osc1.frequency);
osc1.connect(page.audioCX.destination);
What this does is oscillate the pitch of the sound created by osc1. I can change the color to the frequency of osc1 by using osc1.frequency.value, but this doesn't factor in the changes applied by the other parts.
How can I get the resultant frequency from those chained elements?

You have to do the addition yourself (osc1.frequency.value + output of gain).
The best current (but see below) way to get access to the output of gain is probably to use a ScriptProcessorNode. You can just use the last sample from each buffer passed to the ScriptProcessorNode, and set the buffer size based on how frequently you want to update the color.
(Note on ScriptProcessorNode: There is a bug in Chrome and Safari that makes ScriptProcessorNode not work if it doesn't have at least one output channel. You'll probably have to create it with one input and one output, have it send all zeros to the output, and connect it to the destination, to get it to work.)
Near-future answer: You can also try using an AnalyserNode, but under the current spec, the time domain data can only be read from an AnalyserNode as bytes, which means the floating point samples are being converted to be in the range [0, 255] in some unspecified way (probably scaling the range [-1, 1] to [0, 255], so the values you need would be clipped). The latest draft spec includes a getFloatTimeDomainData method, which is probably your cleanest solution. It seems to have already been implemented in Chrome, but not Firefox, as far as I can tell.

Upsampling Audio PCM-data in Javascript with WebAudioApi

For a project, I'm retrieving a live audio stream via WebSockets from a Java server. On the server, I'm processing the samples in 16Bit/8000hz/mono in the form of 8-bit signed byte values (with two bytes making up one sample). On the browser, however, the lowest supported samplerate is 22050 hz. So my idea was to "simply" upsample the existing 8000 to 32000 hz, which is supported and seems to me like an easy calculation.
So far, I've tried linear upsampling and cosine interpolation, but both didn't work. In addition to sounding really distorted, the first one also added some clicking noises. I might also have trouble with the WebAudioAPI in Chrome, but at least the sound is playing and is barely recognizable as what it should be. So I guess no codec- or endianess-problem.
Here's the complete code that gets executed when a binary packet with sound data is received. I'm creating new buffers and buffersources all the time for the sake of simplicity (yeah, no good for performance). data is an ArrayBuffer. First, I'm converting the samples to Float, then I'm upsampling.
//endianess-aware buffer view
var bufferView=new DataView(data),
//the audio buffer to set for output
buffer=_audioContext.createBuffer(1,640,32000),
//reference to underlying buffer array
buf=buffer.getChannelData(0),
floatBuffer8000=new Float32Array(160);
//16Bit => Float
for(var i=0,j=null;i<160;i++){
j=bufferView.getInt16(i*2,false);
floatBuffer8000[i]=(j>0)?j/32767:j/-32767;
}
//convert 8000 => 32000
var point1,point2,point3,point4,mu=0.2,mu2=(1-Math.cos(mu*Math.PI))/2;
for(var i=0,j=0;i<160;i++){
//index for dst buffer
j=i*4;
//the points to interpolate between
point1=floatBuffer8000[i];
point2=(i<159)?floatBuffer8000[i+1]:point1;
point3=(i<158)?floatBuffer8000[i+2]:point1;
point4=(i<157)?floatBuffer8000[i+3]:point1;
//interpolate
point2=(point1*(1-mu2)+point2*mu2);
point3=(point2*(1-mu2)+point3*mu2);
point4=(point3*(1-mu2)+point4*mu2);
//put data into buffer
buf[j]=point1;
buf[j+1]=point2;
buf[j+2]=point3;
buf[j+3]=point4;
}
//playback
var node=_audioContext.createBufferSource(0);
node.buffer=buffer;
node.connect(_audioContext.destination);
node.noteOn(_audioContext.currentTime);

Finally found a solution for this. The conversion from 16Bit to Float is wrong, it just needs to be
floatBuffer8000[i]=j/32767.0;
Also, feeding the API with a lot of small samples doesn't work well, so you need to buffer some samples and play them together.

Render image from audio

Is there a possibility to render an visualization of an audio file?
Maybe with SoundManager2 / Canvas / HTML5 Audio?
Do you know some technics?
I want to create something like this:

You have a tone of samples and tutorials here : http://www.html5rocks.com/en/tutorials/#webaudio
For the moment it work in the last Chrome and the last last Firefox (Opera ?).
Demos : http://www.chromeexperiments.com/tag/audio/
To do it now, for all visitors of a web site, you can check SoundManagerV2.js who pass through a flash "proxy" to access audio data http://www.schillmania.com/projects/soundmanager2/demo/api/ (They already work on the HTML5 audio engine, to release it as soon as majors browsers implement it)
Up to you for drawing in a canvas 3 differents audio data : WaveForm, Equalizer and Peak.
soundManager.defaultOptions.whileplaying = function() { // AUDIO analyzer !!!
$document.trigger({ // DISPATCH ALL DATA RELATIVE TO AUDIO STREAM // AUDIO ANALYZER
type : 'musicLoader:whileplaying',
sound : {
position : this.position, // In milliseconds
duration : this.duration,
waveformDataLeft : this.waveformData.left, // Array of 256 floating-point (three decimal place) values from -1 to 1
waveformDataRight: this.waveformData.right,
eqDataLeft : this.eqData.left, // Containing two arrays of 256 floating-point (three decimal place) values from 0 to 1
eqDataRight : this.eqData.right, // ... , the result of an FFT on the waveform data. Can be used to draw a spectrum (frequency range)
peakDataLeft : this.peakData.left, // Floating-point values ranging from 0 to 1, indicating "peak" (volume) level
peakDataRight : this.peakData.right
}
});
};
With HTML5 you can get :
var freqByteData = new Uint8Array(analyser.frequencyBinCount);
var timeByteData = new Uint8Array(analyser.frequencyBinCount);
function onaudioprocess() {
analyser.getByteFrequencyData(freqByteData);
analyser.getByteTimeDomainData(timeByteData);
/* draw your canvas */
}
Time to work ! ;)

Run samples through an FFT, and then display the energy within a given range of frequencies as the height of the graph at a given point. You'll normally want the frequency ranges going from around 20 Hz at the left to roughly the sampling rate/2 at the right (or 20 KHz if the sampling rate exceeds 40 KHz).
I'm not so sure about doing this in JavaScript though. Don't get me wrong: JavaScript is perfectly capable of implementing an FFT -- but I'm not at all sure about doing it in real time. OTOH, for user viewing, you can get by with around 5-10 updates per second, which is likely to be a considerably easier target to reach. For example, 20 ms of samples updated every 200 ms might be halfway reasonable to hope for, though I certainly can't guarantee that you'll be able to keep up with that.

http://ajaxian.com/archives/amazing-audio-sampling-in-javascript-with-firefox
Check out the source code to see how they're visualizing the audio

This isn't possible yet except by fetching the audio as binary data and unpacking the MP3 (not JavaScript's forte), or maybe by using Java or Flash to extract the bits of information you need (it seems possible but it also seems like more headache than I personally would want to take on).
But you might be interested in Dave Humphrey's audio experiments, which include some cool visualization stuff. He's doing this by making modifications to the browser source code and recompiling it, so this is obviously not a realistic solution for you. But those experiments could lead to new features being added to the <audio> element in the future.

For this you would need to do a Fourier transform (look for FFT) which will be slow in javascript, and not possible in realtime at present.
If you really want to do this in the browser, I would suggest doing it in java/silverlight, since they deliver the fastest number crunching speed in the browser.