Simple room acoustics problems, like room modes, might be solved without shifting the
problems to monetary conflicts if you play music from audiophile player software in your
computer.
Let's see how that works.
In my video " Loudspeaker placement, the long version" I explain that two groups
of acoustical problems might arise when placing speakers in a room: problems below roughly
200 Hz and problems roughly above 600 Hz.
The first group has to do with the size of the room and low frequency wave lengths.
Where half the wave length equals the distance between two room boundaries, we speak of a
room mode.
As soon as a wave with such a length is produced by your stereo, it will cause a resonance
in the room.
When room treatment is done, often so called bass traps are used.
These are a kind of black holes for the lower part of the frequency band: sound that enters
the bass trap will be converted into another form of energy, like heath.
But they are not always practical and often not aesthetically pleasing.
So what if we don't have our stereo play the room mode frequencies and it's first
few harmonics?
I have tried this using the Roon parametric equaliser and it worked quite well.
Other player software has equalisers aboard and thus can be used too, like JRiver Media
Center 24 and Audirvana 3+.
As always with room correction you must place your loudspeakers as good as you can before
you even consider room correction.
Again, my two step approach to loudspeaker placement video tells you exactly how to do
this.
The link is in the show notes.
Also do realise that corrections you make on your listening position might only be appropriate
for a small area around that position and might not work at other listening positions
since standing waves have a position bound behaviour.
The goal here is to only filter out those frequencies that would agitate the room modes,
so the differences per place in the room might be smaller than when equalisation over the
lower 500 Hz is used as room correction software does.
This is how I went about it.
I installed a simple third octave graphics analyser on my smartphone and monitored a
number of tracks that contain a fair amount of low end.
Over these tracks the 125 Hz bar often peaked higher than the neighbouring bars.
If I would only have monitored just one track, the peaking could be the fingerprint of that
recording.
By monitoring several - clearly different - tracks this is avoided.
So now I know that somewhere around 125 Hz there is a resonance but I want more precision.
Making high precision measurements on low frequencies in a room is rather complex but
since room modes are the consequence of the spacing of walls, I calculated the frequencies.
That can be done since we know that the speed of sound, which is about 334 meters or 1,100
feet per second in dry air, at sea level and at 20 degrees celsius or 68 Fahrenheit.
In other words, in the 'standard climate' as used in aviation but usually doesn't
occur on many places on earth.
I say this to prevent you calculating frequencies with three decimals just to be exact.
Rounding at integers doesn't do harm.
About the speed of sound: Some use 331.2 meter per second as speed - which seems to be scientifically
correct, others use 333.3 meters per second since that would give 1 kilometre per 3 seconds
and I have used 334 meters per second in the past so I will consequently keep using that.
The differences are only of academic importance since in most instances climate conditions
will differ anyway.
The ground floor of my house is almost completely open with the exception of the hall and the
cupboard below the stairs.
Half way there is a room divider that for low frequencies is acoustically transparent.
So acoustically the 'room' is 10.6 by 5.4 meters.
To find out the frequencies that belong to these measures some complex math is needed
for we need to divide the speed of sound by the length in meters - or feet if you prefer,
I will use meters.
So, brace yourself, here it comes: the speed of sound is 334 meters and the length of the
ground floor is 10.6 meters.
So we divide 334 by 10.6 to find 31.509434339 Hz, which we round of to 32 Hz.
We do the same for the 5.4 meters: 334 divided by 5.4 is 61,8518519 Hz which we round of
to 62 Hz.
Room modes start working at half the wave length so in my case 16 and 31 Hz. 16 Hz is
so low that I don't expect problems there but the full wavelength is 32 Hz which is
about equal to 31 Hz.
And the harmonics then are 62, 64, 93, 96, 124 and 128 Hz.
I decided to filter 32, 63 95 and 126 Hz and play with the width of the filter.
The 126 Hz must have been the problem I have seen on the third octave analyser and I will
try the lower two to see what impact they have.
I will use these frequencies in the examples below but you should of course use the problem
frequencies of your room.
To filter out a very small part of the frequency spectrum we use the parametric equaliser.
It has this name since all parameters that define a filter can be varied: center frequency,
gain and Q-factor.
The center frequency is the frequency that is attenuated or amplified the most, the gain
defines the amount of attenuation of amplification and the Q-factor defines the width of the
filter.
It might be clear that we want to maintain as much low frequency energy as possible and
only want to loose the problematic frequencies.
So we are going to use a very narrow filter and thus a very high Q-factor.
Now, let's see how we do this in three software players: Roon, Audirvana 3+ and JRiver Media
Center 24.
After starting up Roon, click the volume control in the lower right corner and click DSP.
Select Parametric EQ in the left column.
If it isn't present, first click Add filter in the lower left corner and select Parametric
EQ.
Now you see four peak/dip filters set at zero dB gain.
Click on the pencil of the first filter and choose Band stop, set the frequency at 32
Hz or your home , of course frequency for your and the Q factor at 15.
Do the same for the next three filters but set them for the three higher frequencies
, 63, 95 and 126 Hz in my case.
You can experiment with more filters at higher multiples but it brought me no further improvement.
Now the filters are set we want to go to Headroom management.
I set the Show Clipping Indicator to on and set the Headroom Adjustment to -3.
This is needed to give the DSP function room to calculate without clipping.
The clipping indicator is integrated in the Signal Path light that will turn red when
clipping occurs.
If that happens, just lower the Headroom Adjustment (dB) setting with one or two dB's until
the clipping goes away.
In Audirvana go to Preferences and select Audio Units.
Here choose AUParametric EQ and click Configuration.
Now drag the white dot to the right frequency or alternatively click on the frequency and
change it to 32 Hz.
Also change the gain to -20 dB - which is the maximum setting, change the Q into 15
and click apply.
Repeat this for three more filters, using the three frequencies you found, in my case
63, 95 and 126 as frequency, or, rather those frequencies that give problems in your room,
of course.
Click the Use Audio Units Effect checkbox and you're set.
In Media Center open the menu Player and select DSP Studio.
You can also use Command D on a Mac or Control-D in Windows.
Click Add and select Parametric Equaliser in the left column - not Room Correction - and
select Adjust a Frequency.
In the lower part of the window enter 32 at Frequency, 15 at Q and -40 at Gain.
Leave left, right as is.
Click Add again and enter 63 Hz as frequency, 15 as Q and -40 as gain and do this again
for the other two frequencies.
Of course, you need to enter the problem frequencies of your room.
Now the filters are set, listen to it for a week or so.
Then switch them off again and listen if the filters improved the sound quality.
It is important to get used to 'the new sound' before judging it since our ears
correct for colouration to a certain degree and removing room resonances will decrease
the level of low frequency energy to the proper level.
But our ears - or rather our auditory system - might initially experience it as 'lacking
bass'.
So by listening to the new sound for a week, that will be reset.
After a week listen not to the timbre but to the details you can hear in both situations.
So, not how much bass is heard but how precise you can follow the the bass lines of the instruments.
If it doesn't work, switch off all filters nut the one closest to the dominant resonance
you saw on the third active analyser and start playing with that one.
Vary the frequency sightly or play with the Q-factor.
If that improves the sound, use the same Q-factor for the other filters and use integer division
of the dominant frequency.
When in doubt, you could switch off three of the four filters, reduce the playback level
on your amp and choose an amplification of 10 dB in the dominant frequency filter.
Then vary the frequency until you hear a strong increase in level.
By boosting you agitate that frequency easily but therefore make sure you play at a lowlevel.
If that doesn't work well, temporarily lower the Q-factor to -say - 5.
After finding the resonance frequency, set the gain back to the initial attenuation and
increase the Q-factor to regain the fullness without exiting the room resonance.
Again, when ready set the other filters to the same Q-factor and use integer division
of the frequency used in the dominant frequency filter.
This simple and cheap method will usually yield lower results than professional acousticians
can achieve.
Even if you would have the same education professional acousticians have, you would
lack the experience.
For despite all the computer models they can use, like ray tracing software, it appears
that experience is eminent.
And that doesn't come cheap.
But if your room and speakers pose little problems, this is a fine and cheap solution.
You might loose some quality due to the signal processing.
How much is easily tested by disabling the filters and do a -0.1 dB filter at a low frequency
and with a high Q. Then compare the bypass situation to the filtered situation.
A -0.1 dB high-Q filter will be inaudible to our hearing so if there is a degradation
of sound, it is due to the DSP algorithms.
Room correction takes time to do well and that's not different in this situation.
And you are on your own, I can't help you for each and every situation is different
and I simply can't spare the time to give you personal support.
But I am researching automated systems, like Dirac in the NAD receiver I described already.
So if that's more your thing, subscribe to this channel, or follow me on social media.
If you liked this video, please consider supporting the channel through Patreon or Paypal.
Any financial support is much appreciated.
The links are in the comments below this video in Youtube.
Help me to help even more people enjoy music at home by telling your friends on the web
about this channel.
I am Hans Beekhuyzen, thank you for watching and see you in the next show or on theHBproject.com.
And whatever you do, enjoy the music.
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