Time Domain correction explained

Hello,

I am very curious to know more about how Audyssey corrects the time domain, which is said to be one of its advantages compared to other equalizing techniques.

I am not 100% into the math of FIR filters, but as far as I've read, it is possible to create filters without any kind of phase shifts compared to the theoretical phase distiotion of high Q IIR filters. If we take aside the number of correction points and the frequency response, what exactly does Audyssey do compared to conventional EQ? Can it take the delay as a function of frequency and correct the arrival of frequencies according to a measurement which includes the time domain?

Lets say you have much reverb time at 40-60hz in your room. Apart from correcting any dips/peaks like a conventional EQ, how does Audyssey handle this? Sorry if I sound sceptical, but it does not make sense to me, that it is able to handle reflections in the room. When the sound at a certain frequency "leaves" the speaker, Audyssey cannot do anything to it. So reflections caused by this sound, is not in the hands of Audyssey anymore. The only way to lower these reflections must be to lower the output of the certain frequency coming from the speaker.

I hope my question makes sense, and I'm looking forward to an answer.

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3 Comments

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    Chris Kyriakakis

    Hi Jonas,

    You are right that there is no way to control the sound after it leaves the speaker.  However, we can measure the effects of the reflections that arrive after the direct sound by looking at the time domain response.  It has a certain pattern to it that will depend on the time of arrival of those reflections.  Based on that pattern and the similarity of patterns across multiple measured locations we can identify the problems caused by reflections.  Then, a filter is created to invert those problems as best as possible.

    The key is to not think of the filtering in the time domain.  It's not like a graphic equalizer that can only raise or lower the amplitude at certain frequencies.  Our filters are in the form of impulse responses that operate on the audio signal through an operation called convolution.  As such, they are affecting the signal in each channel in the time domain--hence the name.  An additional benefit of such filters (also called FIR) is that they operate in the time-frequency domain as well.  That means they can be used effectively to lower the ring down time of room modes in the low frequency range.

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    Jonas Hansen

    Hi Chris.

    Thanks for your answer! I will try to look further into it to understand exactly what you're sating.

    I hope its okay that I bring up the EQ systems of your competitors, but I looked into the JBL Synthesis eq systems (The SDECs) which is a $10.000-$15.000 dollar solution. It looks like, JBL is "only" using standard IIR PEQ filters to satisfy a given target curve. In regards to what youre saying about the advantages of FIR filters, this seems weird to me.

    If we put some very good speakers into an acoustically very good room, how big would the need for time domain correction be?

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    Chris Kyriakakis

    The need for time domain based room correction really has nothing to do with how good the speaker is.  Even the perfect speaker will experience reflections from surfaces in the room and these will cause problems that have to be corrected.

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