Web Audio API

How to transmit data over sound

Robert Rypuła

2018.05.17

Presentation overview:

 

  • What is sound wave and how to store it?
  • Web Audio API overview
  • Generate sound - speakers
  • Hear the sound - microphone
  • AnalyserNode
  • Modulation techniques
  • Spectral Waterfall
  • Physical Layer
  • Data Link Layer
  • Where to find more?

What is sound wave and
how to store it
?

Speed of sound: 320 - 350 m/s (~ 1200 km/h)
Radio wave is almost milion times faster!

  • 50 Hz --> 6.7 m
  • 1 000 Hz --> 33.5 cm
  • 15 000 Hz --> 2.2 cm

Sound wave in air travels as local changes in pressure

Wave length examples:

Speaker and mic needs to have moving part

Human hearing range: 20 Hz - 20 kHz

FM radio:  30 Hz - 15 kHz

Text

Whistling: 1 kHz - 2.5 kHz

Enough for voice: 300 Hz - 3400 Hz

Web Audio API + mobile phone: 150 Hz - 6000 Hz

0 Hz

10 kHz

20 kHz

Common Sampling Rates:

Text

  • 44.1 kHz
  • 48.0 kHz (almost all phones)

 

Common sample value precision: 16 bit signed integer

Range: -32768 do 32767

In Web Audio API it's normalized to: -1, +1

How to store sound wave?

Take a sample in constant interval

Text

Humans hear up to 20 kHz!

Why we use such high sampling rate?

Sampling frequency needs to be at least 2x higher than maximum frequency in the signal

44.1 kHz / 2 = 22 050 Hz (2 050 Hz for filtering)

48.0 kHz / 2 = 24 000 Hz (4 000 Hz for filtering)

0 Hz

20 kHz

44.1 kHz

48 kHz

?

Text

Text

first 3 frames
out of 10

light line - what we want to store

dark line - what we will recover

fake frequencies

real frequencies

Text

Sample rate assumed in those two slides: 10 Hz

Maximum frequency: 5 Hz

Web Audio API
overview

Audio Nodes live inside Audio Context

 

 

 

 

 

Very elastic - we are just connecting nodes as we like

AudioContext

AudioContext

Input nodes:

  • OscillatorNode
    Example: generates waves
  • AudioBufferSourceNode
    Example: reading audio files
  • MediaStreamAudioSourceNode
    Example: access microphone stream

Output node:

  • AudioDestinationNode
    Example: final node that represents speakers

~

10111

00101

Effect nodes:

  • BiquadFilterNode
    Filters: lowpass, high pass, bandpass
  • DelayNode
    Add delay to the stream
  • GainNode
    Volume control

 

Visualization/processing nodes:

  • AnalyserNode
    gives time/frequency domain data
  • ScriptProcessorNode
    gives arrays of subsequent samples

DSP

DSP

DSP

FFT

freqDomain

timeDomain

your code

Generate sound - speakers

OscillatorNode

oscNode = audioContext.createOscillator();

GainNode

gainNode = audioContext.createGain();

AudioDestinationNode

adNode = audioContext.destination;

connect()

connect()

AudioContext

audioContext = new AudioContext();

oscillatorNode = audioContext.createOscillator();
gainNode = audioContext.createGain();

oscillatorNode.start();   // <-- required!

oscillatorNode.connect(gainNode);
gainNode.connect(audioContext.destination);
<input 
  type="range" min="0" max="20000" value="440" 
  onChange="oscillatorNode.frequency
    .setValueAtTime(this.value, audioContext.currentTime)"
/>
<input 
  type="range" min="0" max="1" value="1" step="0.01" 
  onChange="gainNode.gain
    .setValueAtTime(this.value, audioContext.currentTime)" 
/>

Hear the sound - microphone

To access microphone we need to have
user permission

Only websites served
via https can access microphone

MediaStreamSourceNode

micNode = audioContext.createMediaStreamSource(stream);

GainNode

gainNode = audioContext.createGain();

AudioDestinationNode

adNode = audioContext.destination;

connect()

connect()

AudioContext

navigator.mediaDevices.getUserMedia(constraints)
  .then(function (stream) {   });
var microphoneNode;  // <-- IMPORTANT, declare variable outside promise

function connectMicrophoneTo(audioNode) {
  var constraints = {
    video: false,
    audio: true
  };

  navigator.mediaDevices.getUserMedia(constraints)
    .then(function (stream) {
      microphoneNode = audioContext.createMediaStreamSource(stream);
      microphoneNode.connect(audioNode);
    })
    .catch(function (error) {
      alert(error);
    });
}

// ...

function init() {
  audioContext = new AudioContext();
  gainNode = audioContext.createGain();

  connectMicrophoneTo(gainNode);
  gainNode.connect(audioContext.destination);

  // ...
}

WARNING:

Microphone connected to speakers. Audio feedback

may occur!!

AnalyserNode

Allows to look at the same audio signal in

two different ways

Frequency Domain:

'fftSize' values

'fftSize/2' values

Time domain:

Here it is!!

It looks like signal has one dominant frequency 

0 dB

-20 dB

-40 dB

-60 dB

-80 dB

-100 dB

-120 dB

'0.5*fftSize' frequency bins

Frequency domain:

AnalyserNode performs Discrete Fourier Transform (Fast Fourier Transform, FFT)

'X' time domain samples are transformed into 'X/2' frequency bins

Frequency bin is expressed in decibels: every 20 dB we have 10x difference in amplitude

MediaStreamSourceNode

micNode = audioContext.createMediaStreamSource(stream);

AnalyserNode

analyserNode = audioContext.createAnalyser();

connect()

AudioContext

navigator.mediaDevices.getUserMedia(constraints)
  .then(function (stream) {   });
function init() {
  audioContext = new AudioContext();

  analyserNode = audioContext.createAnalyser();
  analyserNode.fftSize = 1024;          // <--- fftSize, only powers of two!
  analyserNode.smoothingTimeConstant = 0.8; 

  connectMicrophoneTo(analyserNode);
}

function getTimeDomainData() {
  var data = new Float32Array(analyserNode.fftSize);

  analyserNode.getFloatTimeDomainData(data);

  return data;        // ----> array length: fftSize
}

function getFrequencyData() {
  var data = new Float32Array(analyserNode.frequencyBinCount);

  analyserNode.getFloatFrequencyData(data);

  return data;        // ----> array length: fftSize / 2
}

How to use Analyser Node

  • better resolution -> bigger fftSize
  • bigger fftSize -> more time domain samples 
  • more time domain samples -> longer it takes to collect them

fftSize

fftSize

interval length

function getTimeDomainDuration() {
  return analyserNode.fftSize / audioContext.sampleRate; // [ms]
}

function getFftResolution() {
  return audioContext.sampleRate / analyserNode.fftSize; // [Hz/freqBin]
}

function getFrequency(fftBinIndex) {
  return fftBinIndex * getFftResolution(); // [Hz]
}

Frequency domain output interpretation
is based on sampleRate as well

My tests:

Slowest mobile device was able to deliver 4 unique frequency domain outputs per second.

 

The best fit is fftSize = 8192

Discrete Fourier Transform sandbox ;)

Modulation techniques

Let's assume we want to send 4 different symbols

 

 

How to use wave properties to send them?

?

Amplitude-Shift Keying

Good only for few symbols

 

With more symbols they will be to

close to each other

Phase-Shift Keying

Phase data not available in Web Audio API

 

Not so many symbols per carrier wave

 

Requires custom-made DSP code

Frequency-Shift Keying

Out of the box with Analyser Node

 

More symbols?

  • Expand the bandwidth!
  • Increase FFT resulution!

Spectral Waterfall

How to show frequency domain changes in one row?

Add colors!

0 dB

-40 dB

-80 dB

-120 dB

Desktop vs mobile

 

Most mobile devices blocks all frequencies above 8 kHz

Skip factor 3

Skip factor 5

How to make single frequency more visible?

Merge frequency bins by finding max in each group

Where to allocate our symbols?

0 Hz

10 kHz

20 kHz

filtered out by most mobile phones

too low for mobile phone speaker

~800 frequency bins

Assume fftSize equal 8192

skip factor 3 (3 bins merged into 1)

~265 frequency bins

One byte 

Summary

  • Modulation: Frequency-Shift Keying
  • FFT size: 8192
  • Window duration: 186 ms or 171 ms
  • Smoothing: disabled
  • Freq. bins skiping: 3 merged into 1
  • Bandwidth: from ~1.5 kHz to ~6.0 kHz
  • Symbols: ~256 (1 byte per each FFT)
  • Raw symbol rate: 4 FFTs/s

Works on most devices and browsers

Physical Layer

We can get 4 FFTs per second, each containing 1 byte

Unfortunately useful symbol rate will be only 2 bytes / s

H

e

l

l

o

Why?

Rx1

Rx2

0 s

0.5 s

1.0 s

1.5 s

2.0 s

2.5 s

Rx3

Tx

Synchronization process

Sync in

progress

Waiting

for sync

Sync OK

0 s

1 s

2 s

3 s

4 s

Goal:

 

Find "sync" sequence

with highest

signal strength

Physical Layer synchronization example:

0 Hz

10 kHz

20 kHz

It takes 5.5 seconds to send "hello world"

ASCII range

256 data symbols + 2 sync symbols

More Physical Layer examples:

Data Link Layer

Physical Layer is not checking errors

Solution - pack chunks of data into frames!

Payload

Header

Checksum

Fletcher-8 checksum
sum0 = (sum0 + halfOfByte) % 0x0F;
sum1 = (sum1 + sum0) % 0x0F;

Header

"marker"

Header

length

Data Link example:

Where to find more?

Part 1/3

 

Discrete Fourier Tranform

Part 2/3

 

Web Audio API

Part 3/3

 

Self-made network stack

If you liked it please give

a star on Github here

Audio Network

project website

Thanks for watching!

the end

Web Audio API - Data transmission over sound

By Robert Rypuła

Web Audio API - Data transmission over sound

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