Remove high frequency sound from streaming audio node js

Question

I have a small app that accepts incoming audio stream from the internet and I'm trying to find the frequency of a tone or continuous beep. At the time of the tone / beep it is the only thing that would be playing. The rest of the audio is either silence or talking. I'm using the node-pitchfinder npm module to find the tone and when I use a sample audio clip I made of 2,000Hz the app prints out the frequency within one or two Hz. When I pull the audio stream online I keep getting results like 17,000 Hz. My guess is that there is some "noise" in the audio signal and that's what the node-pitchfinder module is picking up.

Is there any way I can filter out that noise in real time to get an accurate frequency?

The streaming audio file is: http://relay.broadcastify.com/fq85hty701gnm4z.mp3

Code below:

const fs = require('fs');
const fsa = require('fs-extra');
const Lame     = require('lame');
const Speaker  = require('speaker');
const Volume   = require('pcm-volume');
const Analyser = require('audio-analyser')
const request  = require('request')
const Chunker  = require('stream-chunker');
const { YIN } = require('node-pitchfinder')
const detectPitch = YIN({ sampleRate: 44100})
//const BUFSIZE  = 64;
const BUFSIZE  = 500;

var decoder   = new Lame.Decoder(); 
decoder.on('format', function(format){onFormat(format)});

var chunker  = Chunker(BUFSIZE);
chunker.pipe(decoder);

var options = {
    url: 'http://relay.broadcastify.com/fq85hty701gnm4z.mp3',
    headers: {
        "Upgrade-Insecure-Requests": 1,
        "Accept": "text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8",
        "User-Agent": "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14) AppleWebKit/605.1.15 (KHTML, like Gecko) Version/12.0 Safari/605.1.15"
    }
}

var audio_stream = request(options);
//var audio_stream = fs.createReadStream('./2000.mp3');

audio_stream.pipe(chunker);

function onFormat(format)
{
    //if (volume == "undefined")
    volume = 1.0;

    vol      = new Volume(volume);
    speaker  = new Speaker(format);

    analyser = createAnalyser(format);
    analyser.on('data', sample);

    console.log(format);
    vol.pipe(speaker);  
    vol.pipe(analyser); 
    decoder.pipe(vol);
    vol.setVolume(volume);
}

function createAnalyser(format)
{
    return new Analyser({
        fftSize: 8,
            bufferSize: BUFSIZE,
            'pcm-stream': {
            channels: format.channels,
            sampleRate: format.sampleRate,
            bitDepth: format.bitDepth
        }
    });
}

var logFile = 'log.txt';
var logOptions = {flag: 'a'};

function sample()
{
    if (analyser) {

        const frequency = detectPitch(analyser._data)
        console.log(frequency)
    }
}

My goal is to find the most dominant audio frequency in a chunk of data so I can figure out the tone.

I found some code that supposedly does this with python

def getFreq( pkt ):
    #Use FFT to determine the peak frequency of the last chunk
    thefreq = 0

    if len(pkt) == bufferSize*swidth:
        indata = np.array(wave.struct.unpack("%dh"%(len(pkt)/swidth), pkt))*window

        # filter out everything outside of our bandpass Hz
        bp = np.fft.rfft(indata)
        minFilterBin = (bandPass[0]/(sampleRate/bufferSize)) + 1
        maxFilterBin = (bandPass[1]/(sampleRate/bufferSize)) - 1
        for i in range(len(bp)):
            if i < minFilterBin: 
                bp[i] = 0
            if i > maxFilterBin: 
                bp[i] = 0

        # Take the fft and square each value
        fftData = abs(bp)**2

        # find the maximum
        which = fftData[1:].argmax() + 1

        # Compute the magnitude of the sample we found
        dB = 10*np.log10(1e-20+abs(bp[which]))
        #avgdB = 10*np.log10(1e-20+abs(bp[which - 10:which + 10].mean()))

        if dB >= minDbLevel:
            # use quadratic interpolation around the max
            if which != len(fftData)-1:
                warnings.simplefilter("error")
                try:
                    y0, y1, y2 = np.log(fftData[which-1:which+2:])
                    x1 = (y2 - y0) * .5 / (2 * y1 - y2 - y0)
                except RuntimeWarning:
                    return(-1)
                # find the frequency and output it
                warnings.simplefilter("always")
                thefreq = (which + x1) * sampleRate/bufferSize
            else:
                thefreq = which * sampleRate/bufferSize
        else:
            thefreq = -1
        return(thefreq)

Marucci

When I connect to the link you provided in your question I get the following error The file you requested could not be found. Can you provide a downloadable mp3 file of the stream you would like to analyse? – Transilient

@Transilient It looks like the website changes the URL from time to time... this one is currently working: relay.broadcastify.com/m14td6p27nghf8r.mp3 If that doesn't work you can find the link to the stream here: broadcastify.com/listen/feed/14082 (It's silent most of the time). An example of a "tone" would be: edispatches.com/orgs/general/… – Marucci

I downloaded the mp3 and I saw that the sample rate of the mp3 is 11025Hz (but in the code above the sample rate of the pitch detector is set to 44100Hz YIN({ sampleRate: 44100})). If I adjust the sample rate to 11025Hz then I get an output of around 2200 Hz. Would that be the expected result? – Transilient

@Transilient the downloaded file is just a sample of what the tones sound like. Ultimately I'll be getting the data from the streaming mp3 file which has the 44100 sample rate. sorry for the confusion – Marucci

When I open the streaming mp3 file in VLC I get a sample rate of 22050. I recommend to initialize the pitch detection algorithm in onFormat with the sample rate from the stream. – Transilient

Answer 1

Original answer:

I can not provide you with a solution but (hopefully) give you enough advice to solve the problem.

I would recommend that you save a part of the stream you want to analyze to a file and then take a look at the file with a spectrum analyzer (e.g. with Audacity). This allows you to determine if the 17kHz signal is present in the audio stream.

If the 17 kHz signal is present in the audio stream then you can filter the audio stream with a low pass filter (e.g. audio-biquad with type lowpass and frequency at somewhere above 2 kHz).

If the 17 kHz signal is not present in the audio then you could try to increase the buffer size BUFSIZE (currently set to 500 in your code). In the example on node-pitchfinder's GitHub page they use a complete audio file for pitch detection. Depending on how the pitch detection algorithm is implemented the result might be different for larger chunks of audio data (i.e. a few seconds) compared to very short chunks (500 samples is around 11 ms at sample rate 44100). Start with a large value for BUFSIZE (e.g. 44100 -> 1 second) and see if it makes a difference.

Explanation of the python code: The code uses FFT (fast fourier transform) to find out which frequencies are present in the audio signal and then searches for the frequency with the highest value. This usually works well for simple signals like a 2 kHz sine wave. You could use dsp.js which provides an FFT implementation if you want to implement it in javascript. However, it is quite a challenge to get this right without some knowledge of digital signal processing theory.

As a side note: the YIN algorithm does not use FFT, it is based on autocorrelation.

Update

The following script uses the fft data of audio-analyser and searches for the maximum frequency. This approach is very basic and only works well for signals where just one frequency is very dominant. The YIN algorithm is much better suited for pitch detection than this example.

const fs = require('fs');
const Lame = require('lame');
const Analyser = require('audio-analyser')
const Chunker = require('stream-chunker');

var analyser;
var fftSize = 4096;

var decoder = new Lame.Decoder();
decoder.on('format', format => {
    analyser = createAnalyser(format);
    decoder.pipe(analyser);
    analyser.on('data', processSamples);
    console.log(format);
});

var chunker = Chunker(fftSize);
var audio_stream = fs.createReadStream('./sine.mp3');

audio_stream.pipe(chunker);
chunker.pipe(decoder);

function createAnalyser(format) {
    return new Analyser({
        fftSize: fftSize,
        frequencyBinCount: fftSize / 2,
        sampleRate: format.sampleRate,
        channels: format.channels,
        bitDepth: format.bitDepth
    });
}

function processSamples() {
    if (analyser) {
        var fftData = new Uint8Array(analyser.frequencyBinCount);
        analyser.getByteFrequencyData(fftData);

        var maxBin = fftData.indexOf(Math.max(...fftData));
        var thefreq = maxBin * analyser.sampleRate / analyser.fftSize;

        console.log(maxBin + " " + thefreq);
    }
}

Transilient

Thanks for the info. I don't see a 17kHz in the audio file. I'll try increasing the buffer size. I thought FFT doesn't give you distinct frequencies which is why I moved away from trying that. – Marucci

the code I have uses FFT and returns that data: npmjs.com/package/audio-analyser Is there a way I can use that information to get what I'm looking for? Similar to how the python code is working? – Marucci

See the updated answer for an example which uses FFT. However I think that you should use the YIN algorithm instead because it is much more robust than the basic FFT approach. – Transilient