[sdiy] [synth-diy] lock-in amplifiers

Ian Fritz ijfritz at comcast.net
Sun Jan 26 18:36:32 CET 2014


I worked with lockins in the lab for many, many years.  Nothing to do with 
Fourier analysis.  They are used to measure response functions of systems 
with weak (noisy) signals, or with an unwanted dc 
component.  Example:  To  measure the response of a photo sensor, you 
excite the sensor with chopped light and feed a reference signal from the 
chopper into the lockin along with the detected photosignal.  The lockin 
acts as a synchronous detector, ie it detects the component of the input 
signal in sync with the chopper.  It does this by inverting the detected 
signal when the chopper is blocking the input beam and averaging the 
resulting signal with a lowpass filter.  The filter time constant is 
adjustable up to tens of seconds to effectively provide narrow band 
synchronous detection at the chopper frequency.  To measure the spectral 
response of the photo sensor, you scan the wavelength of the beam while 
recording the lockin output.  Any dc leakage current through the photo 
sensor is eliminated, as are noise components except exactly at the 
reference frequency.  There are other uses, but this example is typical.

Ian


At 09:33 AM 1/26/2014, Tom Wiltshire wrote:
>I think what you're describing might actually be fourier analysis looked 
>at a different way.
>
>You multiply by a specific frequency (your "precision frequency and 
>precision multipliers") and then you can get the amplitude for that frequency.
>
>
>On 26 Jan 2014, at 16:14, "cheater00 ." <cheater00 at gmail.com> wrote:
>
> > On Sun, Jan 26, 2014 at 4:50 PM, cheater00 . <cheater00 at gmail.com> wrote:
> >> Hi guys,
> >> I've recently stumbled upon the concept of a lock-in amplifier.
> >> Basically, it is a way to recover an AM signal from a carrier, even if
> >> the carrier is buried deep in noise.
> >>
> >> what you get is something like a filter that's got a Q of about 10000.
> >> A normal band pass of Q=100 is considered absurdly steep.
> >>
> >> This helps in two ways. For one thing, you can uncover this carrier
> >> frequency from within a huge amount of noise.
> >>
> >> Also, it's much easier to build a precise oscillator than it is to
> >> build a precise high-Q filter. You can do it in digital without being
> >> penalized by terrible things like aliasing or decimation.
> >>
> >> Cheers,
> >> D.
> >
> > A natural application for this would be a resonator. So you recover
> > the amplitude envelope of a specific, very narrow frequency, within a
> > complex sound. Because of what was said in the previous email (the
> > low-pass filter needs to be able to easily disjoin the near-0
> > frequencies from any frequencies derived from where f0!=f), the
> > amplitude envelope needs to be much slower than the frequency in
> > question.
> >
> > If you need to be able to do quick amplitudes at low frequencies, you
> > might need to up-convert the audio first, so that what was e.g. 20Hz
> > is now 20020 Hz - and therefore even an amplitude envelope at 19 Hz is
> > going to be easy to recover.. at least in theory.
> >
> > Anyways, what this approach gives you is that you don't need to use
> > steep Q filters any more. Instead, you use a precision frequency and
> > precision multipliers. Then, you can use the recovered amplitude
> > envelope to modulate a sine wave at that frequency, or at another
> > frequency (e.g. an octave up), interval (e.g. using the frequency + a
> > 6th is pretty cool), or a chord, and the waveforms can be sine waves,
> > or other waves.
> >
> > Since none of the oscillators need to change pitch or amplitude
> > abruptly, it's possible to create them in digital in such a way that
> > makes them sound as pure as analog waves. Amplitude modulation and
> > mixing should be done in analog. You need a lot of DACs, but they can
> > be low quality, since bit depth necessarily matter for repetitive
> > waves. This means you can use codec or multi-output DAC chips which
> > are inexpensive.
> >
> > Cheers,
> > D.
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