[WoodyLuvr]

There’s some theory behind the Meier crossover linked in the description, but I’m not sure if that’s what you’re looking for. In regards to guidelines for use, it’s hard to say because everyone has their own preference and style of music. At 100, you are hearing full mono essentially, but in my experience the crossfeed squashes the sense of space as low as 15. Others might prefer something higher. There’s no way to recommend a universal setting.

Actually, your basic description above helped me a lot! "...At 100, you are hearing full mono essentially..."

I feel like I missed a crossfeed plugin somewhere. Did somebody mention they were using one other than Meier or BS2B?

Both @ironmine and @castleofargh mentioned both Foo VST xfeed and 112 dB Redline Monitor in these threads here, here, and here.

Foo VST xfeed (aka foo_dsp_xfeed)

112 dB Redline Monitor

I tried them both and they worked (behaved) very well I simply just couldn't hear a major improvement over the simpler Meier Crossfeed plugin so I went back to that. I will give 112 dB Redline Monitor another try again in the coming days to double check if I may have missed something.

 

[sonitus mirus]

Equalizer APO is outstanding. Load it, set it, save it, and done. It uses 1% of one of my laptop's cpu cores. I made a simple reduction in the treble frequencies that, by my ears, made a difference that has me extremely excited with the results.

Using the 31-band EQ setting, I adjusted the following parameters to subtly remove the harsh sounds I was hearing with some music when raising the volume quite a bit to "jam" to a section of a song.
2KHz (-1dB)
2.5KHz (-2dB)
3.15KHz (-2dB)
4KHz (-3dB)
5KHz (-3dB)
6.3KHz (-4dB)
8KHz (-3dB)
10KHz (-3dB)
12.5KHz (-2dB)
16KHz (-1dB) (fyi...I can't hear 16KHz unless I significantly raise the volume with a test tone in a quiet room to 100-106dB; and I rarely play any music this loudly and only for perhaps 15-20 seconds, if even that long)

My speakers are horn-loaded Klipsch RP-280F, and the relatively large horn/tweeter section produces plenty of treble. I sit close to them, too, so taming this frequency range has made a tremendous improvement at louder volumes. There is a very slight compromise at very low volume listening levels, but then I am typically playing music in the background at that volume. The overall result is a significant win.

 

[castleofargh]

I'm getting close, but I just can't seem to grasp how to use the VST host to play music through MathAudio Room EQ.

Because I use a DAC connected to either USB or optical, I have to use some virtual cable solution to channel the sound through any software.

I use the VB-Audio Hi-Fi cable input as the default playback device with a sample rate of 44100 and the recording device as VB-Audio Hi-Fi cable output.



With this setup I can stream my music with a browser or via a desktop player.

I was able start a 64-bit VST host, use the Umik-1 microphone as the input, chain MathAudio EQ as a VST plugin, copy the mic's calibration.txt into the MathAudio EQ folder, and I was able to take room measurements with the frequency sweep and saved a .snr file.

Where I get lost is how to play music back in step 11 of this guide.

http://mathaudio.com/room-eq.htm

In the VST host, I can select a chain that seems to detect music being played, I can hear the music over the speakers, but the MathAudio Room EQ plugin does not seem to do anything to the sound when I adjust the sliders or switch back and forth from Room EQ and Bypass.



I've tried various input and output setting in the VST host. I've had echos, silence, and what appears to be a little static that can be heard if I adjust the volume on my DAC's preamp, but nothing I change on the MathAudio plugin seems to do anything.

I don't think the "listen to this device" thingy should be used at all. the purpose of that stuff is to bounce whatever input directly to the speakers, which is not what you want here. the virtual cable already links what it needs. I'd redo the measurements just in case you ended up somehow measuring the test signal + the sound picked up by the mic and sent back to the speakers at the same time(probably that echo/Larsen stuff you got at some point).

 

[ironmine]

No you can't, although it depends on what you mean by "similar". Room correction is a bit of a misleading term because what you end up with is not a "correct" room, it's a somewhat improved response but not a correct response. We can improve a room's response with EQ but EQ only addresses the total energy at a range of frequencies, not the other main factor of time.

I was not talking about using a simple EQ in the computer for DRC. I meant "REW+Convolver" or "MathAudio Room EQ".

Whatever sound manipulations miniDSP is doing digitally inside its blackbox, it can be done, for free, in your computer.

 

[sonitus mirus]

I don't think the "listen to this device" thingy should be used at all. the purpose of that stuff is to bounce whatever input directly to the speakers, which is not what you want here. the virtual cable already links what it needs. I'd redo the measurements just in case you ended up somehow measuring the test signal + the sound picked up by the mic and sent back to the speakers at the same time(probably that echo/Larsen stuff you got at some point).

I think I did try just about every combination, including unchecking the "Listen to this device" option. When I made the room measurement, I had the microphone as the input and the frequency sweep sounded right. I'm only having a problem attempting to find a way to play music that goes through the MathAudio Room EQ.

I'm sure I could get it to work with Foobar, but then I only have two CDs ripped on the computer now, and I would get bored after a couple of days of listening to the same stuff, no matter how awesome it sounded. I wish everything was as simple as getting fat.

 

[ironmine]

Some of the older folk I’ve met simply wouldn’t be up to the task, and that’s not to say anything of their intelligence, because these guys used to build DIY amps. They’re just from a different time. Room correction DSP, if it were a custom software process only, would effectively leave some very devoted people out.

What can be so difficult about buying an USB mic, plugging it into the PC, taking several measurements with the mic in the Room Measurement mode of MathAudio Room EQ, then switching to the Room EQ mode and listening to the results while fine tuning the position of the slider?

If "old folks" cannot handle these simple steps, they won't handle the miniDSP box either, because I am sure the procedure of configuring the miniDSP is similar at best (and even more difficult)..

 

[castleofargh]

Actually, your basic description above helped me a lot! "...At 100, you are hearing full mono essentially..."


Both @ironmine and @castleofargh mentioned both Foo VST xfeed and 112 dB Redline Monitor in these threads here, here, and here.
I tried them both and they worked (behaved) very well I simply just couldn't hear a major improvement over the simpler Meier Crossfeed plugin so I went back to that. I will give 112 dB Redline Monitor another try again in the coming days to double check if I may have missed something.

the one slider does all on the latest Meier crossfeed is obviously simple, but what is it doing when we move it? I have no idea. crossfeed involves taking one channel and sending it to the other channel with the level/signature altered and with a given delay. both of which should ideally comply with your own head for optimal result. so either you get lucky with Meier, or you'd probably get a more fitting result with something offering a little more customization. of course with the second case it's like when using EQ, there is the matter of being able to set up stuff yourself by ear. last time I tried to do that on a crossfeed plugin, I think I just went with music sent to one channel and moving the settings around until the sound seemed to come from the direction of my speaker for that channel. the rest for "fine tuning" was purely taste based.

I think I did try just about every combination, including unchecking the "Listen to this device" option. When I made the room measurement, I had the microphone as the input and the frequency sweep sounded right. I'm only having a problem attempting to find a way to play music that goes through the MathAudio Room EQ.

I'm sure I could get it to work with Foobar, but then I only have two CDs ripped on the computer now, and I would get bored after a couple of days of listening to the same stuff, no matter how awesome it sounded. I wish everything was as simple as getting fat.

is it only mathaudio or the VST host you can't get working? if you put another VST in it, does it affect the sound? I see the level meter on each module picking a signal, does this signal change with music or does it look like it's some noise staying at a fixed level from whatever issue in some settings?
sorry for being captain obvious but I don't have more to offer right now ^_^.

 

[sonitus mirus]

is it only mathaudio or the VST host you can't get working? if you put another VST in it, does it affect the sound? I see the level meter on each module picking a signal, does this signal change with music or does it look like it's some noise staying at a fixed level from whatever issue in some settings?
sorry for being captain obvious but I don't have more to offer right now ^_^.

My problem is trying to figure out what to do in steps 7-12 of this guide: http://mathaudio.com/room-eq.htm

I can get the VST host to play streaming music that is output to my DAC. I see the level meter moving and hear sound, and when I pause the music, the level meter drops to zero. Still, I was never able to hear any differences when I applied changes as outlined in step 11 and 12, no matter how drastic the changes or how loudly I was playing the music. Granted, I have a big amp and large speakers in a relatively small room, so I am a bit hesitant to just move sliders that control volume levels willy-nilly, but I hear no differences at any extreme setting.

 

[Strangelove424]

Actually, your basic description above helped me a lot! "...At 100, you are hearing full mono essentially..."


Both @ironmine and @castleofargh mentioned both Foo VST xfeed and 112 dB Redline Monitor in these threads here, here, and here.

Foo VST xfeed (aka foo_dsp_xfeed)

112 dB Redline Monitor

I tried them both and they worked (behaved) very well I simply just couldn't hear a major improvement over the simpler Meier Crossfeed plugin so I went back to that. I will give 112 dB Redline Monitor another try again in the coming days to double check if I may have missed something.

Thanks! That's extremely helpful. I'll do a new round an editing to the op soon.

 

[Strangelove424]

What can be so difficult about buying an USB mic, plugging it into the PC, taking several measurements with the mic in the Room Measurement mode of MathAudio Room EQ, then switching to the Room EQ mode and listening to the results while fine tuning the position of the slider?

If "old folks" cannot handle these simple steps, they won't handle the miniDSP box either, because I am sure the procedure of configuring the miniDSP is similar at best (and even more difficult)..

I have older friends, and in my opinion the software route would be too difficult for them without a person there to help. I will not be going into detail about them or their capabilities, as I don't feel that's a matter for forum discussion. The great thing about discussing all these options for DSP is being aware that they even exist. Whether one chooses to use one or the other is a personal choice.

 

[gregorio]

Whatever sound manipulations miniDSP is doing digitally inside its blackbox, it can be done, for free, in your computer.

What sound manipulations is miniDSP doing? Unless you can answer that question you do not know if you can do the same in your computer, let alone for free! Actually, you can do the same thing in your computer as Dirac Live has a software version for MAC/PC but it's not free, it's $770! I have not tried MathAudio software but I have compared (not blind) an Audyssey consumer system with Dirac Live and REW+Convolver. The Dirac was the best and significantly better than REW+Convolver. I've also heard some pro-audio digital room correctors but not in direct comparison. None of these systems are doing exactly the same thing or in the same way, some of them are relatively simple and some are very sophisticated. All of which brings me back to my opening statement, "it depends on what you mean by "similar"".

G

NB: I'm not an expert in this field and my determination of "best" and "significantly better" is only my subjective opinion.

 

[old tech]

What can be so difficult about buying an USB mic, plugging it into the PC, taking several measurements with the mic in the Room Measurement mode of MathAudio Room EQ, then switching to the Room EQ mode and listening to the results while fine tuning the position of the slider?

If "old folks" cannot handle these simple steps, they won't handle the miniDSP box either, because I am sure the procedure of configuring the miniDSP is similar at best (and even more difficult)..

As one of the old folks, perhaps explain how your solution works in a listening room with a playback chain that does not include a PC or laptop? My files are on a server played wirelessly through a streamer. So would I then need to put a laptop in the chain and boot it up everytime I want to listen to music, how inconvenient (I certainly don't want to tamper with the actual files) Then there is the point raised by Gregorio, are the free software implementations as good as something like the miniDSP?

 

[ironmine]

As one of the old folks, perhaps explain how your solution works in a listening room with a playback chain that does not include a PC or laptop? My files are on a server played wirelessly through a streamer. So would I then need to put a laptop in the chain and boot it up everytime I want to listen to music, how inconvenient (I certainly don't want to tamper with the actual files) Then there is the point raised by Gregorio, are the free software implementations as good as something like the miniDSP?

Well, I am 42 years old, so I may be "old folks" myself to some young people!

Please note that if you decide you use a miniDSP, you still need to connect an USB mic to a laptop or PC to run initial measurements.

Did you see the note: "Windows /Mac computer required to run measurement/calibration software" at the website and "To configure your Dirac Series audio processor, you will require a Windows or Apple Mac computer" in the manual?

Secondly, in my opinion, a streamer is as useless as a miniDSP. Please excuse my straightforwardness, but I don't understand how people can spend money on it. Any cheap laptop can "stream" files wirelessly through a network from a server, another PC in the network, etc. Frankly speaking, I don't even get why you need to "stream" anything. Why not just keep your audio files on a hard drive inside your PC or laptop? Or external USB drive?

It's a brilliant marketing idea to scare "old folks" with the "complexity of using a computer for audio" and sell, at expensive prices, to them single-functional devices (such as streamers and miniDSP and data storage servers, etc.) whose sole purpose is to do something so simple that any cheap laptop or PC can do easily, for free, and on a more advanced level. Plus, with computers, you are much more flexible and future-proof.

Thirdly, it only takes 10 seconds to wake up even an old model laptop (with a regular hard drive) from the state of hibernation. If you have a modern laptop which has not a regular hard drive, but a SSD inside, it will be even faster. So activating your laptop does not take longer than switching your amps or DAC.

(As for me, I don't use a laptop for music, I have a desktop computer and it's running 24 hours per day, it's silent). My audio files are on a 2TB external hard disk and the folder where the music is stored is shared through the network. My media player in the other room can play it. My laptop in the kitchen can play it. My wife's PC in the other room can play it.)

Why do you think that miniDSP implementation is better? Correcting "time smearing" is marketing bull, you cannot directly control the duration of a bass resonance with active room correction (i.e. DSP). The best such active (digital) RC can do it to reduce the amplitude (level) of a resonating peak. Of course, if you reduce the level at the frequency which resonates in your room, this resonating frequency will die faster, but it's indirect control. If you want to directly control the resonance duration, you need not active RC, but passive RC (diffusion and absorption panels, bass traps, SBIR and RFZ panels, etc.). Most people are, of course, to lazy to make, buy and install such panels.

Either active or passive room correction alone will not help much, only the combination of both approaches will give you a good result.

 

[bigshot]

If you play everything through a media server and it's just 2 channel, I don't see any advantage to not using a software EQ through the computer. If you have a multichannel or Atmos system, or if you have multiple sources in your system, the computer becomes less convenient.

And whether or not to install panels in a room depends on the materials used to make the room and the particular acoustics. In my case, the whole room is 1950s knotty pine. I can't cover that up with acoustic panels. It would look terrible. Every situation involves tradeoffs and compromises.

 

[old tech]

Well, I am 42 years old, so I may be "old folks" myself to some young people!

I'm 12 years older than you so I'm probably in the gramps territory.

Please note that if you decide you use a miniDSP, you still need to connect an USB mic to a laptop or PC to run initial measurements.

Did you see the note: "Windows /Mac computer required to run measurement/calibration software" at the website and "To configure your Dirac Series audio processor, you will require a Windows or Apple Mac computer" in the manual?

Yes I know that. It is a one-off measurement. There is no need to keep a PC or laptop in my listening room after that.

Secondly, in my opinion, a streamer is as useless as a miniDSP. Please excuse my straightforwardness, but I don't understand how people can spend money on it. Any cheap laptop can "stream" files wirelessly through a network from a server, another PC in the network, etc. Frankly speaking, I don't even get why you need to "stream" anything. Why not just keep your audio files on a hard drive inside your PC or laptop? Or external USB drive?

Are you sure you mean streaming rather than a streamer? Sure a laptop or PC can stream music wirelessly (just as a NAS or a wireless hard drive can) but you still need a device to convert the wireless signal into a digital or analog wired signal for the amp or pre-amp. Please don't tell me you are using Bluetooth. Likewise your comment about keeping the audio files on a PC or laptop, why is that different or better than having the files on a NAS particularly when the NAS is also used as a back up for all the family's PCs and laptop. Either way, you still need a device (a streamer...) to convert that wireless signal to something your stereo can use.

It's a brilliant marketing idea to scare "old folks" with the "complexity of using a computer for audio" and sell, at expensive prices, to them single-functional devices (such as streamers and miniDSP and data storage servers, etc.) which do something that any cheap laptop or PC can do easily, for free, and on a more advanced level.

Sure, there is marketing behind nearly all products and services in a modern economy - not sure where you are getting this scaring old folks bit - but understand people value many attributes of a product or service apart from its ability to perform its core purpose. That is the reason we all don't drive Toyota Camry's (and they would be incredibly cheap if all cars were Camrys) or live in Soviet style apartments. People pay more money for Apple products, rather than a generic implementation, simply because it works with limited fuss - that is my attraction with using a NAS/streamer combination and perhaps something like the miniDSP.

Thirdly, it only takes 10 seconds to wake up even an old model laptop (with a regular hard drive) from the state of hibernation. If you have a modern laptop which has not a regular hard drive, but a SSD inside, it will be even faster. So activating your laptop does not take longer than switching your amps or DAC.

I think you are still missing the point. How is that laptop going to communicate with the hi fi? Sure you can use a media player but then essentially you are using a streamer but now need to lug around a laptop rather than controlling your music through a phone or a small tablet. The alternative of keeping the laptop connected to the stereo via USB or other wired connections has absolutely no appeal to me, it would be almost as inconvenient as playing a record on my TT.

(As for me, I don't use a laptop for music, I have a desktop computer and it's running 24 hours per day, it's silent). My audio files are on a 2TB external hard disk and the folder where the music is stored is shared through the network. My media player in the other room can play it. My laptop in the kitchen can play it. My wife's PC in the other room can play it.)

Again, what is so different to using a NAS which also backs up all the PCs and laptops? Is your media player in the other room wireless? If so, it is essentially a streamer. If it is wired into your network, that is hardly high tech and not suitable for my purposes.

Why do you think that miniDSP implementation is better? Correcting "time smearing" is marketing bull****, you cannot directly control the duration of a bass resonance with active room correction (i.e. DSP). The best such active (digital) RC can do it to reduce the amplitude (level) of a resonating peak. Of course, if you reduce the level at the frequency which resonates in your room, this resonating frequency will die faster, but it's indirect control. If you want to directly control the resonance duration, you need not active RC, but passive RC (diffusion and absorption panels, bass traps, SBIR and RFZ panels, etc.). Most people are, of course, to lazy to make, buy and install such panels.

Either active or passive room correction alone will not help much, only the combination of both approaches will give you a good result.

I don't know if the miniDSP implementation is better or not, hence the earlier question. I like its potential for convenience, sitting passively between the streamer and amp (after the initial setting up) which can then be forgotten as an integrated component of the playback chain. Btw, I agree that DSPs are no substitute for proper room treatment and my listening room does have basic treatment (to the extent that it is still aesthetically pleasing and acceptable to the wife). It is more a curiosity around what DSPs can do to further enhance transparency. I've got a lot to learn in that regard but as the acoustics of my set up are quite good, it is not something I'm willing to sacrifice convenience for.