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Article: Audiolense Digital Loudspeaker and Room Correction Software Walkthrough


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Hi @mitchco,

Thank you for your article and ebook. They have been very helpful. I've redone my 2.1 with digital XO for the sub and tweeter using audiolense. I'm very happy with the results.

Unfortunately, I've only managed to get measurements using windows mme or directsound (I do click 'using separate playing and record streams'). Bernt said ideally ASIO4all, or Wasapi should be used. Unfortunately, I always get an error message saying 'perhaps too many channels', or PaInvalidChannelCount, unexpected host error etc... I've tried AL 5.5, &6.0 both in 32 bit and 64 bit. I've installed Jriver, activated WDM, etc but having no luck.

I must be missing something, or maybe I'm worrying about nothing and should stick with what works?

Do you have any suggestions?

Omid

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For measuring, with audiolense 5.5 (32 bit), my laptop runs on Windows 10, 64 bit. My input is a Calibrated USB mic (UMM6). The output is a Xonar U5 external sound card, set to 5.1, 48kHz, 24 bit (both windows and the sound card are set to the same sample rate). The sound card's front output goes to the analog input of a benchmark DAC2 --> AHB2 --> Tannoy Stirling GR woofer. The R+L rear outputs go to the analog input of a single velodyne DD12 subwoofer (all equalizers, crossover defeated). The Center+Subwoofer outputs go to the analog input of a marantz NR1609--> Tannoy Stirling GR tweeter. 

The channels in AL are set to 4,5 for L+R subwoofer, 0,1 for woofers, 2,3 for tweeters, which works correctly when the driver is winmme, or directsound. Just can't get it to work with JRiver, Wasapi or ASIO4A.

For playback, I use Roon Rock and its DSP. USB output to Benchmark DAC -->woofers. Internal roon rock soundcard -->subwoofer. Xonar soundcard-->marantz--> tweeter. I then use Adobe audition to measure by how many milliseconds  the signal is delayed for each output and correct for that. I play sweep test tones through Roon, and measure with REW. I'm getting a decent step response and house curve matching frequency response. After DSP correction, the volume does drop by 10dB or so, which is annoying (I use partial correction, only correcting up to 9000Hz, so as to avoid a larger drop). 

Please let me know if I left out any info you need.

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Hi @omid Very nice! If you are getting a good step and frequency response with REW, then everything is fine.  Does the Xonar card come with an ASIO driver? One thing you could try if you have not already is to check the response of the signal path sans speakers and mic. But if you are getting a good step and frequency response out of the speakers, then it is likely there are no issues at all.

 

Wrt the attenuation of the digital filter, in JRiver's convolution you could click on normalize filter volume which would bring some gain back. Is there a similar setting in Roon? Another way is to add a digital gain stage. In JRiver, one could use the eq plugin just to add gain, but with no eq. There are VST plugins that can do this as well. I have used Blue Cat's Gain Suite to do this. Just have to watch for any clipping.

 

Seems like you have it all sorted. I think Bernt is working on some of the multichannel issues in AL V6, so this may also allow the use of Wasapi or ASIO. But it seems to me there is nothing to worry about.

Cheers,

Mitch

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Thank you. That helps. 

I moved my subwoofer yesterday, and tried to repeat the measurements. The first set of sweeps worked. When I tried another sweep, the sound stuttered (buffer under run?). After that, I started getting error messages PaDeviceNotFound or something like that (Puzzling since I hadn’t touched the driver or any settings). Tried numerous pc restarts and relaunchings of  AL, same result. 

I’ll try safe mode today, maybe there is a driver conflict. I was just curious if you knew of any tricks to sort out my problems. 

Wrt volume, Roon does let you amplify. It also has a clipping warning. Although the signal is diminished by 10dB or so, the signal clips if I amplify it (i guess some frequencies are boosted by more than 10dB).  

Benchmark DACs and amplifiers don’t have a lot of gain, so 10dB matters. 

I’ll eventually get rid of the xonar soundcard and marantz for the tweeter and get another DAC and amplifier (or maybe I’ll finish building my Nelson pass F5 v3). You’re lucky your Hilo can simultaneously process bass, mid and treble streams at the same time. I technically need 3 DACs. 

 

Cheers. 

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I’m having a puzzling problem, I’m hoping someone with DSP knowledge (hint, @mitchco) may help me solve. 

 

I made a convolution filter through AL making sure ‘all in one’ is selected (so R and L channel corrections are correlated). My mic was equidistant within 1/8 inch when I recorded and generated the correction.

 

I then implemented the convolution in Roon and recorded the output from the sound card directly into my laptop’s line in. The sound card handles the twitters (ie 1800Hz and above).  I recorded the signal using adobe audition. R and L channel signals start and stop at the same time without any DSP, whereas filter corrected sounds have 1 ms delay on the R channel (34 cm difference). The delay is not frequency dependent, ie not likely to be a phase correction related delay.  

 

I imagine the twitter signal should arrive at one’s ears at exactly the same time for a centered voice. Am I missing something, or is Audiolense tripping?

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  • 1 month later...

No worries. I've been through so many iterations that I can't recall what the issue was at the time. I have figured it out though. 

I'm just going to summarize a few of my findings so far in case it helps anyone:

Audiolense has some bugs/driver incompatibilities that are frustrating (and frankly should not be there for the price of the software). Bernt is thankful very helpful.

ASIO4ALL comes with a optional extra software that I hadn't downloaded. The extra software is needed if you have a USB microphone, make sure to download it. Set the buffer to 2048 bits.

'Check speakers' is good for channel mapping but it has bugs. Play with winmme directsound etc until it works for mapping the channels. Once you've done so, you don't need to check again. You can switch the driver to ASIO4ALL after that. 

Playing sweeps of more than 10 seconds does weird things, and results in alarming knocking sounds. I think it's due to clock drift. Once that happens, reset the delays for your speakers to 0, or you'll keep on getting weird knocking and distortions during your sweeps.

The low and high frequency of the target curve needs to carefully follow your speaker's measured curve or your step response looks weird. Also the more dB correction is needed to match your target curve, the lower your response curve will be. The DSP correction can easily attenuate your signal by 10 to 20 dB. Make sure your pre-amp/amp have enough dB gain (and/or your speakers are sensitive enough) to achieve a decent maximum loudness upon playback.

Although the option to' detect and correct polarity' was checked off, looking at my sub's response, I realized it was inverted. Once I switched the polarity, my step response improved a lot.

 

 

 

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On 9/29/2019 at 6:02 PM, omid said:

Hi @mitchco,

Thank you for your article and ebook. They have been very helpful. I've redone my 2.1 with digital XO for the sub and tweeter using audiolense. I'm very happy with the results.

Unfortunately, I've only managed to get measurements using windows mme or directsound (I do click 'using separate playing and record streams'). Bernt said ideally ASIO4all, or Wasapi should be used. Unfortunately, I always get an error message saying 'perhaps too many channels', or PaInvalidChannelCount, unexpected host error etc... I've tried AL 5.5, &6.0 both in 32 bit and 64 bit. I've installed Jriver, activated WDM, etc but having no luck.

I must be missing something, or maybe I'm worrying about nothing and should stick with what works?

Do you have any suggestions?

Omid

I have the same issue. Bernt answered on the forum that there are lots of issues using various ASIO drivers. He also said MME is fine if you set the PC sound card to the bit depth/sample rate  you are measuring in and prevent drift; the results should then be indistinguishable from using ASIO. 

Main listening (small home office):

Main setup: Surge protector +>Isol-8 Mini sub Axis Power Strip/Isolation>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three .

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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On 11/17/2019 at 10:24 AM, omid said:

No worries. I've been through so many iterations that I can't recall what the issue was at the time. I have figured it out though. 

I'm just going to summarize a few of my findings so far in case it helps anyone:

Audiolense has some bugs/driver incompatibilities that are frustrating (and frankly should not be there for the price of the software). Bernt is thankful very helpful.

ASIO4ALL comes with a optional extra software that I hadn't downloaded. The extra software is needed if you have a USB microphone, make sure to download it. Set the buffer to 2048 bits.

'Check speakers' is good for channel mapping but it has bugs. Play with winmme directsound etc until it works for mapping the channels. Once you've done so, you don't need to check again. You can switch the driver to ASIO4ALL after that. 

Playing sweeps of more than 10 seconds does weird things, and results in alarming knocking sounds. I think it's due to clock drift. Once that happens, reset the delays for your speakers to 0, or you'll keep on getting weird knocking and distortions during your sweeps.

The low and high frequency of the target curve needs to carefully follow your speaker's measured curve or your step response looks weird. Also the more dB correction is needed to match your target curve, the lower your response curve will be. The DSP correction can easily attenuate your signal by 10 to 20 dB. Make sure your pre-amp/amp have enough dB gain (and/or your speakers are sensitive enough) to achieve a decent maximum loudness upon playback.

Although the option to' detect and correct polarity' was checked off, looking at my sub's response, I realized it was inverted. Once I switched the polarity, my step response improved a lot.

 

 

 

Good points @omid Was there a reason for longer than 10s sweeps? Just curious. Yes, you want to follow the minimum phase response of the roll-off of your speakers. One can also use partial correction if the top end is to one's liking.

 

I would also recommend a small cycle HF window to prevent over correction if using full frequency correction. Typically 6 cycles for the low end and 1 cycle for the top end works pretty well in most rooms without the top end sounding compressed or strident.

 

One could start with 6 dB of correction, gen a filter and then try 12 dB of correction, gen a filter and A/B to see if there is a preference for one or another. For sure, there is filter insertion loss. But one can a) normalize the gain (in JRiver anyway) which brings some of the gain back. b) add digital gain is an option, just watch out for clipping.

 

Interesting about the sub polarity. I have the exact same case in my other article using Audiolense to integrate subs, but I did not invert the polarity and the step response came out perfect. Curious if you have a before and after step response with the polarity switch?

 

How does your system sound? 

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Thank you @mitchco

The longer sweep was to get a better signal/noise ratio (the measurement goes from good to excellent). I realize the extra resolution is unnecessary.

I attached the measured step with neg polarity on the sub. Not sure why it makes a difference (I figured the DSP should easily correct for that, but it didn't). The simulated step response comes out funny (attached).

I'll attach the corrected step response and corresponding step response, and REW. I use 6 dB correction. 12 dB and 18 dB give me similar curves (slightly higher peak on the step response).

I do add some gain back in Roon (just short of clipping).

I'm happy with how it sounds. I've ordered a Motu 8a and a new amp for the tweeters.

inverted step.jpg

inverted stimulated step.jpg

corrected step.jpg

simulated step response corrected.jpg

rew.jpg

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  • 5 months later...

Hi Mitch:

 

I posted on this thread in September 2018, asking if you had put any thought into an optimal nearfield 'sitting at a computer up against a wall' setup? What speakers and DSP configuration would you recommend?

 

With so many of us stuck at home, on our computers all day, I cannot imagine a better time for this :) 

 

Thanks!

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@TooSteep I have had this article half complete for some time and got distracted 🙂 As far as what speakers, this is up to your personal preference. The idea is that Audiolense DSP would correct for the speakers location relative to one's near field listening position. The DSP configuration is a little different as we use a shorter time domain correction window to take care of the reflections off the front wall and desk, but not so much for the room as it is a nearfield setup.

 

I am hoping to finish the 2nd half of the article at some point using KEF LS50's in a nearfield setup, but @Archimago has my amplifiers for an upcoming testing article. Due to Covid, I may not get those back for a while...

 

Cheers,

Mitch

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Hi again @mitchco

 

Incredible article. I'm trying to get my feet wet with digital room EQ, with a basic 2-channel passive crossover speaker setup. The 2-channel amp has PRE-OUT which feeds 2 subs (left and right, behind the speakers). Room is symmetrical. 

 

I thought about Dirac but that's a separate box and not really as flexible for the future as doing it in software. So I am leaning to Audiolense, for exporting the RIFF wav filter files into HQPlayer's convolution engine (I'm a long time HQPlayer user). That will be a nice clean system making use of existing software I already use and without any additional/new boxes. 

 

Some questions if you don't mind!

 

  • Q1. For taking multiple measurements around the listening position, is it a complicated process in Audiolense? Or just a box you tick in the software and it's quite an automated process?

 

  • Q2. Regarding: "The frequency correction is included in all version. This alone is enough to improve the sound quality in the finest audio systems. Our 2.0 offering is in our opinion one of the best bargains in high end hi-fi. The surround version provides frequency correction for all speakers in a surround setup, proper time alignment and a sophisticated crossover solution for seamless bass management. Our high end offering – the Audiolense XO – comes with True Time Domain (TTD). The TTD correction fully synchronizes the first arrival for all frequencies and all speakers involved. Audiolense XO also comes with a very sophisticated and configurable crossover solution that covers all bases."

 

I'm trying to workout, for my simple system (passive speaker crossover and the simple way my subs are connected via amp pre-outs) the difference between 'surround' and 'XO' versions. The former features 'time alignment' but the later has 'TTD'. Kind of related to my other thread yesterday about general terms used in time domain correction, what's the difference here in this particular case with Audiolense?

 

 

image.png.6ece296e2cabe1c85c02c9ed7cde4806.png

 

and here:

 

image.png.0f90802e9e0d759753439cbe2418a2eb.png

 

 

  • Q4. Can you attach the latest version PDF help file / manual for Audiolense? I couldn't find it on the website.

 

  • Q5. Can you recommend USB mic's to use? Will Audiolense work with the  MiniDSP UMIK-1 ?

 

  • Q6. A more advanced query about digital crossover (not something on my radar for a while but I am curious and like learning). With your particular setup, I'm trying to work out how Audiolense is actually able to split frequencies between tweeter and midbass, to your 2-channel Lynx Hilo? I thought the Hilo is stereo and Aurora is their multi-channel version. So I was confused how does Audiolense and/or J River, 'see' the separate tweet and middbass (and their DACs)? I'm probably missing something really obviously.

 

Or do you have 2 x Hilo's? 

 

 

Just some high level questions for now. If I can get comfortable and not feel too daunted, I'll go full throttle and get your eBook and get Audiolense.

 

Thanks again !

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Quick follow-up (can't edit my post above).

 

Would this combo work with Audiolense:

 

 
and
 
 
So the RME will feature both mic input and D-to-A ?
 
I did see Bernt mention in an old post that using the miniDSP UMIK-1 might have time domain correction issues so the above should eliminate that issue:
 
image.thumb.png.c28abe9d173f0db5b701f10e6ef440a0.png
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8 hours ago, asdf1000 said:

Hi again @mitchco

 

Incredible article. I'm trying to get my feet wet with digital room EQ, with a basic 2-channel passive crossover speaker setup. The 2-channel amp has PRE-OUT which feeds 2 subs (left and right, behind the speakers). Room is symmetrical. 

 

I thought about Dirac but that's a separate box and not really as flexible for the future as doing it in software. So I am leaning to Audiolense, for exporting the RIFF wav filter files into HQPlayer's convolution engine (I'm a long time HQPlayer user). That will be a nice clean system making use of existing software I already use and without any additional/new boxes. 

 

Some questions if you don't mind!

 

  • Q1. For taking multiple measurements around the listening position, is it a complicated process in Audiolense? Or just a box you tick in the software and it's quite an automated process?

 

  • Q2. Regarding: "The frequency correction is included in all version. This alone is enough to improve the sound quality in the finest audio systems. Our 2.0 offering is in our opinion one of the best bargains in high end hi-fi. The surround version provides frequency correction for all speakers in a surround setup, proper time alignment and a sophisticated crossover solution for seamless bass management. Our high end offering – the Audiolense XO – comes with True Time Domain (TTD). The TTD correction fully synchronizes the first arrival for all frequencies and all speakers involved. Audiolense XO also comes with a very sophisticated and configurable crossover solution that covers all bases."

 

I'm trying to workout, for my simple system (passive speaker crossover and the simple way my subs are connected via amp pre-outs) the difference between 'surround' and 'XO' versions. The former features 'time alignment' but the later has 'TTD'. Kind of related to my other thread yesterday about general terms used in time domain correction, what's the difference here in this particular case with Audiolense?

 

 

image.png.6ece296e2cabe1c85c02c9ed7cde4806.png

 

and here:

 

image.png.0f90802e9e0d759753439cbe2418a2eb.png

 

 

  • Q4. Can you attach the latest version PDF help file / manual for Audiolense? I couldn't find it on the website.

 

  • Q5. Can you recommend USB mic's to use? Will Audiolense work with the  MiniDSP UMIK-1 ?

 

  • Q6. A more advanced query about digital crossover (not something on my radar for a while but I am curious and like learning). With your particular setup, I'm trying to work out how Audiolense is actually able to split frequencies between tweeter and midbass, to your 2-channel Lynx Hilo? I thought the Hilo is stereo and Aurora is their multi-channel version. So I was confused how does Audiolense and/or J River, 'see' the separate tweet and middbass (and their DACs)? I'm probably missing something really obviously.

 

Or do you have 2 x Hilo's? 

 

 

Just some high level questions for now. If I can get comfortable and not feel too daunted, I'll go full throttle and get your eBook and get Audiolense.

 

Thanks again !

 

To answer your questions.

Q1 - yes, it is straight ahead to do mulitiseat measurements in AL.

Q2 - I highly recommend True Time Domain (TTD) correction with the added bonus to be able to use digital XO's in the future. The surround time alignment, aligns the multiple speakers, but does not do TTD from what I recall. But that is a question for Bernt.

Q3 - thanks Chris!

Q4 - You can download AL 6.6 in demo mode: http://juicehifi.com/download/ and then you can access the PDF help file which will answer many of your questions. When you launch the program, select the XO version.

Q5- The miniDSP USB mic will work just fine. If you join the Audiolense forum: https://groups.google.com/forum/#!forum/audiolense you can search on a test I did with the UMIK-1 USB mic compared to an analog mic. Using Bernt's "clock drift correction" that is in the later versions of AL, works perfect and virtually no difference in timing between the two mics.

Q6 - The Lynx Hilo has 8 digital channels, 6 channels of analog output and 2 channels of analog input. Here is a simple diagram of it working and in this case, I am taking a loopback measurement of my stereo triamp system with the digital XO and correction filters loaded in JRiver's convolution engine. PS. Always use ASIO if you can, and in my case the Hilo's ASIO driver is excellent and also happens to be multi-client, which allows me to do this:

 

Loopback.jpg

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9 hours ago, mitchco said:

 

To answer your questions.

Q1 - yes, it is straight ahead to do mulitiseat measurements in AL.

Q2 - I highly recommend True Time Domain (TTD) correction with the added bonus to be able to use digital XO's in the future. The surround time alignment, aligns the multiple speakers, but does not do TTD from what I recall. But that is a question for Bernt.

Q3 - thanks Chris!

Q4 - You can download AL 6.6 in demo mode: http://juicehifi.com/download/ and then you can access the PDF help file which will answer many of your questions. When you launch the program, select the XO version.

Q5- The miniDSP USB mic will work just fine. If you join the Audiolense forum: https://groups.google.com/forum/#!forum/audiolense you can search on a test I did with the UMIK-1 USB mic compared to an analog mic. Using Bernt's "clock drift correction" that is in the later versions of AL, works perfect and virtually no difference in timing between the two mics.

Q6 - The Lynx Hilo has 8 digital channels, 6 channels of analog output and 2 channels of analog input. Here is a simple diagram of it working and in this case, I am taking a loopback measurement of my stereo triamp system with the digital XO and correction filters loaded in JRiver's convolution engine. PS. Always use ASIO if you can, and in my case the Hilo's ASIO driver is excellent and also happens to be multi-client, which allows me to do this:

 

Loopback.jpg

 

Thanks Mitch!

 

Just a couple more questions:

 

Q1. For a dedicated listening room with just one listening position, would you still recommend multiple measurements. Example, 6 inches around the normal head position? Or is single is sufficient in your experience?

 

Q2. Does your eBook guide the reader through the multiple measurements process in Audiolense E.g. recommended number of measurements around the sweet spot, the spacing distance between measurements around the sweet spot and how to put it all together in Audiolense to get one nice overall filter.

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Q1 - I have used AL with single measurement and multiple measurements, both sound excellent. Personally, I would go with the single measurement. AL has a psychoacoustic filter applied which does not fill in all of dips, so a) it automatically avoids over correction and therefore does not require multiple averaged measurements b) already produces a wide sweet spot.

For example, my single measurement covers my three seat sofa with uniform frequency response, which I have verified with multiple REW measurements after correction. But since the software does both, it is easy enough to try both and let your ears decide.

 

Q2 - While my book uses Acourate as the DSP software (also excellent software!), all of the concepts and procedures apply to Audiolense and any other well designed DSP software. While the actual steps are different, they are performing the same DSP tasks.

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8 hours ago, mitchco said:

Q1 - I have used AL with single measurement and multiple measurements, both sound excellent. Personally, I would go with the single measurement. AL has a psychoacoustic filter applied which does not fill in all of dips, so a) it automatically avoids over correction and therefore does not require multiple averaged measurements b) already produces a wide sweet spot.

For example, my single measurement covers my three seat sofa with uniform frequency response, which I have verified with multiple REW measurements after correction. But since the software does both, it is easy enough to try both and let your ears decide.

 

Q2 - While my book uses Acourate as the DSP software (also excellent software!), all of the concepts and procedures apply to Audiolense and any other well designed DSP software. While the actual steps are different, they are performing the same DSP tasks.

 

Legend, thanks again Mitch.

 

I will start with Audiolense but I can say with certainty that the  'fear of missing out' bug will hit me and I'll end up trying Acourate too - which I will definitely need your book for...

 

Really appreciate all your advice. 

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  • 2 weeks later...

Hi again @mitchco

 

I watched your fantastic recent YouTube video and you mention the best DRC software solutions don't boost any frequencies.

 

But in your screenshots in this article I see the max correction boost is set to +6dB.

 

Would you recommend setting this max boost to 0dB ?

 

 

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8 hours ago, asdf1000 said:

Hi again @mitchco

 

I watched your fantastic recent YouTube video and you mention the best DRC software solutions don't boost any frequencies.

 

But in your screenshots in this article I see the max correction boost is set to +6dB.

 

Would you recommend setting this max boost to 0dB ?

 

 

Hi @asdf1000 

Can you post a link to the video? I'd love to watch it. 

Thanks.

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4 hours ago, omid said:

Hi @asdf1000 

Can you post a link to the video? I'd love to watch it. 

Thanks.

 

Hi @omid

 

Here you go: https://www.youtube.com/watch?v=VOHhvFb2TTs&feature=youtu.be&t=62m19s

 

It's a really really informative interview.

 

@mitchco's comments about correcting peaks and dips are from 1:24:00, "really good DRC software only cuts, never boosts".

 

So I was wondering if this max correction boost setting in Audiolense should be set to 0dB.

 

 

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13 hours ago, asdf1000 said:

Hi again @mitchco

 

I watched your fantastic recent YouTube video and you mention the best DRC software solutions don't boost any frequencies.

 

But in your screenshots in this article I see the max correction boost is set to +6dB.

 

Would you recommend setting this max boost to 0dB ?

 

 

Cheers. Cracking open the Audiolense manual: "The Max boost setting determines how high a dip is allowed to be lifted by the correction filter. A too high setting will consume the dynamic range of the setup. Typical setting is below 12 dB and 6 dB is a good figure most of the time. Higher values work – and sometimes a higher value is needed to get the desired result. The correction boost will also be limited by the measurement and correction window – and quite often to an extent where a 6dB max boost setting hardly makes a difference."

 

Not to be confused with that the correction filter never exceeds 0 dBFS. This is what I was referring to in the video. As soon as you make a correction filter and examine it in AL, you will see what I mean. It is not "boosting" in the traditional sense, like a  +PEQ for example.

 

I think you said you were playing with Audiolense? I would recommend creating a few corrections filters with different so-called boost settings and examine what the correction filter looks. And most importantly listen to the impact of increasing and decreasing this setting and what it does to the dynamic range. There usually is a "sweetspot" for any given loudspeaker/room setup, plus what each persons preference is. 

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