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[Alexx]

Those new speaker settings are WONDERFUL.  This awesome program just keeps getting better and better!

[Matt]

Glad you like it.

After thinking about it some more, I think we're going to incorporate volume correction into the speaker distance algorithm.

If we know your left speaker is 10% farther from you than your right speaker, it's pretty simple math to figure out the decibel difference from this.  I think this should be applied automatically when distance correction is in use instead of requiring the user to dial it in with the Level slider. (the level slider will of course still be available)

Another change I'd like is to add a 'Mute' and 'Solo' button on the Levels section to help with testing.

[Alexx]

Glad you like it.

After thinking about it some more....

All excellent ideas!  I know a lot of people grumble about the price of Media Center, and upgrades and etc., but I can't wait to shell out for the next version.  I'll take it out of the grocery budget if I have to.  Whatever it takes to keep JRiver going, and going strong, like the Energizer Bunny.

[llafriel]

Now this is great stuff! This allows me to have speakercorrection per zone, love it! Could we also have the distance available in metric units?
Ooh! What about enabling drc filters per speaker? This would allow multichannel roomcorrection..
http://www.duffroomcorrection.com/wiki/Main_Page

[jmone]

Glad you like it.

After thinking about it some more, I think we're going to incorporate volume correction into the speaker distance algorithm.

If we know your left speaker is 10% farther from you than your right speaker, it's pretty simple math to figure out the decibel difference from this.  I think this should be applied automatically when distance correction is in use instead of requiring the user to dial it in with the Level slider. (the level slider will of course still be available)

Another change I'd like is to add a 'Mute' and 'Solo' button on the Levels section to help with testing.

Sounds good Matt, couple of observations to back this up:
1) Setting speaker distance is a good start for volume correction if you don't have a SPL meter but it is great to keep the manual DB setting to take account of stuff in the room (eg furniture gets in the way, some speakers may be in corners, some are not etc etc).
2) Mute helps with testing and calibration but you will need a test tone option as well

You pretty well have most of the pre-amp stuff now available in MC.  The only other option I see between MC and mid range Recievers in this area are:
1) Microphone based Auto Tuning (you stick a mic at the listening position, it plays a bunch of test tones in order from each speaker and the adjusts for each channel, polarity, distance, loudness, even EQ).
2) Per channel EQ - I don't use it but some do especially when trying to match the "timbre" of different brand, type, and size of the three speakers across the front sound stage.

[rick.ca]

My RealTek onboard audio includes room correction. How do this and the DSP version interact? Do I need to take care only one is engaged at a time?

[Matt]

2) Mute helps with testing and calibration but you will need a test tone option as well

Maybe next to the channel selection combobox it could have three buttons: "Solo" "Mute" "Tone"

Tone would output pink noise.

The only complication would be when you choose a channel that's not currently in use the buttons would need to disable.

2) Per channel EQ - I don't use it but some do especially when trying to match the "timbre" of different brand, type, and size of the three speakers across the front sound stage.

Equalization is already done per-channel (with separate threads for smooth multi-CPU performance), so this is just a user interface issue.  I'm not really sure the best way to solve it.

A third thing for that list might be convolution-based correction with impulse recordings.  I've never gotten results I consider worthwhile when I've played with this in the past so I'm less eager to hurry into this area.

[Matt]

My RealTek onboard audio includes room correction. How do this and the DSP version interact? Do I need to take care only one is engaged at a time?

You would not want to run two room correction DSPs in the output chain.

This goes for most other effects as well like environment, reverb, equalization, etc.

It's possible some output modes like ASIO or WASAPI exclusive would not allow the driver to apply effects even if they're selected.

An advantage of using driver-level processing is that it'll be used with other programs or games.  But an advantage of using Media Center is you'll get a better audio pipeline with generally higher quality effects that you'll have more control over.

[jmone]

Maybe next to the channel selection combobox it could have three buttons: "Solo" "Mute" "Tone"

Tone would output pink noise.

Sounds like a plan.

The only complication would be when you choose a channel that's not currently in use the buttons would need to disable.

I'd not bother disabling the buttons for now as if it is not in use you would not hear anything anyway so it would not matter much (unless I've missed the point) - or see below for the menu idea

Equalization is already done per-channel (with separate threads for smooth multi-CPU performance), so this is just a user interface issue.  I'm not really sure the best way to solve it.

You could have a Check Box that the user can select for each Ch that is in use, then a Configure Button become active for each CH and that then takes you to a screen with all the settings on it - distance, loudness, eq, test, etc etc - repeat for each Ch.

A third thing for that list might be convolution-based correction with impulse recordings.  I've never gotten results I consider worthwhile when I've played with this in the past so I'm less eager to hurry into this area.

I believe you!  (mostly because I have no idea what this even is but it sounds like it need to be firmly dealt with!)

[rick.ca]

Thanks, Matt. The question in my mind, of course, is what about other applications that don't have their own correction. I think a good enough answer might be if they don't have it, they can do without it. So if it makes an important difference to a game, for example, hopefully there's appropriate corrections settings within the game. Otherwise, I need to remember to change the setting before and after—a daunting task for my tired old brain. Needless to say, I appreciate MC's ability to configure things differently in separate zones.

Afterthought: I suppose I could launch a game with a batch file that changes the driver settings in the registry for the duration of the game.

[)p(]

A third thing for that list might be convolution-based correction with impulse recordings.  I've never gotten results I consider worthwhile when I've played with this in the past so I'm less eager to hurry into this area.

I use a vst based convolver (Voxengo Prestine Space) with mc all the time. I really like the results. I think native support for drc would be a pretty popular audiophile feature.

peter

[gappie]

Thanks, Matt. The question in my mind, of course, is what about other applications that don't have their own correction. I think a good enough answer might be if they don't have it, they can do without it. So if it makes an important difference to a game, for example, hopefully there's appropriate corrections settings within the game. Otherwise, I need to remember to change the setting before and after—a daunting task for my tired old brain. Needless to say, I appreciate MC's ability to configure things differently in separate zones.

Afterthought: I suppose I could launch a game with a batch file that changes the driver settings in the registry for the duration of the game.

i think that the room correction is only working for audio, not for the files using directX, right matt? so when you want to use it with your movies..

 
gab

[mojave]

Glad you like it.

After thinking about it some more, I think we're going to incorporate volume correction into the speaker distance algorithm.

If we know your left speaker is 10% farther from you than your right speaker, it's pretty simple math to figure out the decibel difference from this.  I think this should be applied automatically when distance correction is in use instead of requiring the user to dial it in with the Level slider. (the level slider will of course still be available)

This is a bad idea. Volume correction is used for differences in distance, but also for differences in speaker sensitivity. For example, my center speaker has a sensitivity of 90.5 dB@1m and my mains have a sensitivity of 86 dB@1m. At the same distance, I need to attenuate the center speaker by 4.5 dB in order to match my mains.

Another change I'd like is to add a 'Mute' and 'Solo' button on the Levels section to help with testing.

Great idea!

[Matt]

This is a bad idea. Volume correction is used for differences in distance, but also for differences in speaker sensitivity. For example, my center speaker has a sensitivity of 90.5 dB@1m and my mains have a sensitivity of 86 dB@1m. At the same distance, I need to attenuate the center speaker by 4.5 dB in order to match my mains.

You're of course right that speaker sensitivity needs to be accounted for.

But distance provides a good starting point, and as long as the manual adjustment for level remains, I think it makes a nice addition.  It will be in the next build.

It's kind of neat to sit really close to a speaker and listen to how it sounds almost like mono.  Then dial in the distances and the image neatly balances out.

[mojave]

In the Speakers DSP there should also be the option for Large or Small for each speaker and a setting for the crossover frequency. When set to Large, each speaker gets the full range audio signal with the option to also route the sound below the crossover frequency to the sub. Each speaker should also have a check box for "Bass to Sub Plus Mains" that is available if Large is selected. If it is set to Small each speaker gets the frequencies above the crossover frequency with the frequencies below the crossover sent to the sub and combined with the LFE signal. The crossover should have a High Pass Filter with a 12dB/octave slope and a Low Pass Filter with a selectable slope (12 dB or 24dB/octave slope). This selection could be made on the Subwoofer settings page and should only be allowed when the mains are run as Large and "Bass to Sub Plus Mains" is selected. Each speaker should also have a check box for "Bass to Sub Plus Mains." Some like their bass routed to the mains in addition to the sub. These settings would allow the Speakers DSP to handle bass management and provide greater flexibility.

The reason for the selectable slope of the subwoofer LPF is because some like to listen to music in a 2.1 configuration. They like their mains to run full range and their subs to add in the low frequencies below the natural frequency rolloff of their mains. Because main speakers of different designs roll off at different rates, having an adjustable rolloff for the sub is helpful for arriving at the best speaker/sub integration.

With my 7.1 channel system, for example, I would set the Output Format DSP to "7.1 channels" and surround mixing to "no mixing (unused channels silent)". I would then set each speaker to small and use a crossover of 80 Hz. Now regardless of how many channels I am listening to, MC should route all bass to my subwoofer channel. If I am listening to two channel music, my speakers should get from 80Hz and up and the sub from 80Hz and down.

[Matt]

In the Speakers DSP there should also be the option for Large or Small for each speaker and a setting for the crossover frequency. When set to Large, each speaker gets the full range audio signal with the option to also route the sound below the crossover frequency to the sub. Each speaker should also have a check box for "Bass to Sub Plus Mains" that is available if Large is selected. If it is set to Small each speaker gets the frequencies above the crossover frequency with the frequencies below the crossover sent to the sub and combined with the LFE signal. The crossover should have a High Pass Filter with a 12dB/octave slope and a Low Pass Filter with a selectable slope (12 dB or 24dB/octave slope). This selection could be made on the Subwoofer settings page and should only be allowed when the mains are run as Large and "Bass to Sub Plus Mains" is selected. Each speaker should also have a check box for "Bass to Sub Plus Mains." Some like their bass routed to the mains in addition to the sub. These settings would allow the Speakers DSP to handle bass management and provide greater flexibility.

The reason for the selectable slope of the subwoofer LPF is because some like to listen to music in a 2.1 configuration. They like their mains to run full range and their subs to add in the low frequencies below the natural frequency rolloff of their mains. Because main speakers of different designs roll off at different rates, having an adjustable rolloff for the sub is helpful for arriving at the best speaker/sub integration.

With my 7.1 channel system, for example, I would set the Output Format DSP to "7.1 channels" and surround mixing to "no mixing (unused channels silent)". I would then set each speaker to small and use a crossover of 80 Hz. Now regardless of how many channels I am listening to, MC should route all bass to my subwoofer channel. If I am listening to two channel music, my speakers should get from 80Hz and up and the sub from 80Hz and down.

We're still thinking about this.

A parametric equalizer available per speaker with presets like 'Large Speaker', 'Medium Speaker' and 'Small Speaker' might be one approach.

[mojave]

I have a Behringer DCX2496 that I have been using for bass management. It does parametric equalization and also provides digital crossovers. For each speaker you can set a HP filter and a LP filter with crossover slopes including 6dB, 12dB, 18dB, 24dB, and 48dB per octave with three filter types (Butterworth, Bessel, and Linkwitz-Riley) and using whatever frequency you want. It also has adjustable delays with manual or automatic correction for room temperature and distance. The parametric equalizer allowed for a gain of +/- 15dB, frequency, and Q (bandwidth). The Q (bandwidth setting) is essentially the width in octaves that is affected by the EQ. There is no way I could use this advanced parametric EQ to perform the crossover function. It works in a completely different way. Even using the crossovers in the DCX2496 require a lot of knowledge and measurements to get them set correctly. I have a calibrated measuring microphone which helped greatly.

For most users, being able to select Large or Small and a frequency is the best choice. I'm not sure what a Medium setting would be. You are either using a crossover or not using a crossover. There is no middle ground.  ?

There is disagreement whether all speakers should use the same crossover. For starters, you could have a crossover setting under the Subwoofer page. This would set everything to the same crossover frequency and it could be either on or off.

My current soundcard provides its own crossover setting. However, one would expect audiophile software to provide more choices.  

[Alex B]

You're of course right that speaker sensitivity needs to be accounted for.

But distance provides a good starting point, and as long as the manual adjustment for level remains, I think it makes a nice addition.  It will be in the next build.

It's kind of neat to sit really close to a speaker and listen to how it sounds almost like mono.  Then dial in the distances and the image neatly balances out.

Actually on typical home theater receivers the speaker distance setting adjusts channel specific delay. Here the first article that Google found: http://www.audioholics.com/tweaks/speaker-setup-guidelines/setting-speaker-levels-distance-in-a-surround-sound-system

The channel specific volume adjustments are separate because there can be many factors in addition to the speaker distance. As said, individual speakers can have a different sensitivity and also a different power amp may be used for different speakers. For instance, the L & R speakers may be connected to a separate stereo power amp and the other speakers to a HT receiver. Also room acoustics can have a significant effect to the speaker volume level.

[mojave]

Maybe next to the channel selection combobox it could have three buttons: "Solo" "Mute" "Tone"

Thanks for adding these with the latest released build. However, Tone is only available when audio is playing and then it overlaps the current audio. For testing purposes, it really should only play when audio is not playing.

[Matt]

Thanks for adding these with the latest released build. However, Tone is only available when audio is playing and then it overlaps the current audio. For testing purposes, it really should only play when audio is not playing.

Check 'Solo' and 'Tone' to output to only that speaker.

[mojave]

I found by using the mute and/or solo button that the Room Correction DSP only works when Playback Method is set to Automatic. If I add any DirectShow filters then it does not work. Also, it does not work for movies (I tried with ripped DVD's and MKV's). I have used AC3Filter for movies, but I found out yesterday that I can use AC3Filter with audio files, too. I only tried it with FLAC and APE files, but it worked great for distance settings, delay, crossover, and speaker output configuration (2.1, 4.1, etc). This is what MC's Room Correction needs to be like. In order to use AC3Filter with audio files, in Playback Method I selected JRiver Audio Engine (using DirectShow filter). I then checked AC3Filter.

[Audionut]

Just another vote for native convolver support in mc!!! I use it for room acoustic correction based on the measured response of the system in my room at the listening position, it is a superb feature for crazy audiophiles like me   and hopefully others too.
Great that .VST support is there already, but the integration with the vst plugin and mc is not ideal of course. It is brilliant that it is there, but a native one could be so much better:

A must would be to have a default filter file loaded and enabled for standard playback. A nice to have option would be remote commands to switch the convolver filters like presets (great to compare different target responses).

Thanks!

Joerg

I use a vst based convolver (Voxengo Prestine Space) with mc all the time. I really like the results. I think native support for drc would be a pretty popular audiophile feature.

peter

[Matt]

DSP for movies, DVD, television

I have a feeling this is coming.  Maybe soon.


Convolver

Perhaps someone who is a fan of convolver could help us get something set up in a way that sounds good.

I spent some more time tonight playing with convolution.  I was using a Sennheiser HD600 headphones, but have also used it with my surround sound theater system.  I haven't found anything that seems really interesting.

It feels like an overly complicated way of doing equalization.  Speaker reverb correction might also be neat, although it seems like it's easy to do more harm than good with heavy processing for this.

Thanks for any tips.

[)p(]

I use it for room acoustic correction based on the measured response of the system in my room at the listening position,

Yeah I use it also like that for room correction by measuring the response at the listening position with a decent calibrated mic.

If you have not tried the vst plugin route you should it works 100% ok with mc. You can also use the open source directx drc convolver with mc. The only issue with the dx convolver is that you have to reenable the settings each time you start mc.

peter

[Audionut]

Yes, I use a VST plugin which works very well, but so far every plug in I used had one or the other issue in terms of working without needing some sort of work around. Like you mention, reenabling the setting, or  my current one keeps the default setting, but the automatic level correction (convolution is always reduced in output volume over original signal) does not work when I start mc new on my dedicated PC. I need to enter the DSP Plugin, see the user interface, switch a filter for it to be picked up. Of course I can do this each time, but it is just annoying to have to remember and do it every single day.

Since I have tried so many convolution plugins I have the feeling the only way to get this to work smoothly without manual interaction would be to have the whole process in the hands of one developer, ideally inside mc...

If you have not tried the vst plugin route you should it works 100% ok with mc. You can also use the open source directx drc convolver with mc. The only issue with the dx convolver is that you have to reenable the settings each time you start mc.

peter

[Audionut]

I'd be more than happy to give all the input I have for a working solution  

When you say "played with convolution", can you tell me what you do? Are you using downloadable room response filters or are you measuring a response, creating a target response and then claculating a FIR filter to use in the convolution engine? I guess since you used Headphones this time it is more like an effect plugin to add or remove reverb? Those I would agree, are mostly more entertaining then really useful in the context of regular listening.

I use convolution with a measured response and calculated filter baesd on a target curve for the frequency response, it is quite a journey to get something that sounds excellent. But, if you have a room that has significant bass problems (typically room modes) or less than perfect speaker placement, even a fairly automated correction filter sounds better to me than the orginal unprocessed sound. The software I use for that calculation also provides group delay correction of the speaker signal, so there is additional potential for improvement.

There is some analogy to parametric equalization, just that IMHO it is comparing a butchers knife to a scalpel. To stay in this idea, with convolution you need the tools and experience of a brain surgeon to really optimize the result. Doing this for a 7.1 home theatre setup would most certainly require the "Ueber" Professor, I am glad if I don't mess up my 2 channel "patient"  

I also had very well tuned setups with dedicated rooms where I could not manage to improve the sound. So for me the key to get something that sounds good lies in how much are you able to specify the target response and how much do you know about how to relate this to the sound you hear with the speakers to optimize for the best result.

Convolver

Perhaps someone who is a fan of convolver could help us get something set up in a way that sounds good.

I spent some more time tonight playing with convolution.  I was using a Sennheiser HD600 headphones, but have also used it with my surround sound theater system.  I haven't found anything that seems really interesting.

It feels like an overly complicated way of doing equalization.  Speaker reverb correction might also be neat, although it seems like it's easy to do more harm than good with heavy processing for this.

Thanks for any tips.

[Audionut]

Another experience I have with room correction with convolution filters is that quite often a filter that sounds close to perfect to me is difficult to get used to for others when they hear this kind of room correction the first time. Quite often, they have been listening to a setup that produces a certain sound signature due to the bass resonances that they are now used to. A good filter makes this almost linear and it can be "diasappointing" at first and perceived as "loss of bass", "not as loud or dynamic" etc. Usually in those cases it takes an hour or so to get used to the new sound and then starting to appraciate the positive results. The good results that are achievable which you can listen for when you try it are:

- Bass is not always there booming as one note in the background, it can be abesent, go very deep, have different texture that changes according to each recording. There will still be recordings that sound overly fat and boomy in the bass, but it will be only those that were actually mixed that way
- You don't have to "hunt" or concentrate on the stereo image, it is locked in and very stable. You can distinguish very clearly intermediate location cues like a bit left of center or so and not just left-middle-right (but also with room correction, for this one needs to be in or very close to the famous "sweet spot"
- The general tonal balance of the whole system does not change as much with output volume as without correction (this also confuses people in the beginning). You can turn up the volume more before getting the feeling it is too loud and have to turn it down. The cleaner response in the room makes this possible
- Depth of the image is usally improved. Also the feeling of listening fatigue is removed, sounds tend to be finer and less aggressive (precise cuts rather than jigsaw edges)

Ok, I hope this is somehow making sense to you, it is all coming from my "audiophile" stereo perspective and preferences.

Joerg

[)p(]

The good results that are achievable which you can listen for when you try it are:

Great list...very similar to my own experiences with drc.


peter

[audunth]

I'm also very interested in being able to use DRC FIR filters with MC. I haven't really started setting it up yet (I know once I get started it'll eat up all my free time for at least a few weeks  ), but I've done some measurements using a free program called Room EQ Wizard (http://www.hometheatershack.com/roomeq/) and the results are terrible and in big need of correction. I've done some manual corrections with the parametric eq of my Yamaha receiver, and even those simple corrections makes a huge difference. I'm going to use a program called Audiolense (http://www.juicehifi.com/) which makes it pretty straight forward to set up the correction filters. Now, if it would be just as easy to set up those filters in MC instead of having to use external plugins that don't really work 100%, requiring various workarounds etc. it would be perfect

Those who don't know what proper correction can do to the sound can read this review http://www.avrev.com/home-theater-accessories/acoustics-eq-room-tuning/audyssey-sound-equalizer.html . This $2500 unit uses the same type of FIR filters that we're talking about here.

[flac.rules]

Yeah, but the audessey technology does multiple measurments positions and uses som kind of fuzzy logic to weigh them and create a total responce. It seems a bit more advanced. And room correction seems to be pretty dependant on a good implmentation, I recently saw a blind-test of room correction systems done in harman kardon labs, some scored better than than no correction, one scored the same, and some scored worse.

[audunth]

The newest version of Audiolense also has multi seat correction. I haven't used Audyssey myself, but to me it seems Audiolense is more advanced, since it gives you full control of every aspect of the correction.

[flac.rules]

The newest version of Audiolense also has multi seat correction. I haven't used Audyssey myself, but to me it seems Audiolense is more advanced, since it gives you full control of every aspect of the correction.

That does not follow automatically, although lack of control is a problem with audyssey, it can still use algorithms that are a lot more advanced for its correction.

[mojave]

I read that the Computer Audiophile website will be doing a review of MC14 soon. Their focus is primarily on two channel audio with external DAC's and no subwoofers. However, I think it would help differentiate MC from other media players if it also had advanced subwoofer crossover capability. I have an earlier post in this thread detailing what I would like to see.

[nwboater]

I read that the Computer Audiophile website will be doing a review of MC14 soon. Their focus is primarily on two channel audio with external DAC's and no subwoofers. However, I think it would help differentiate MC from other media players if it also had advanced subwoofer crossover capability. I have an earlier post in this thread detailing what I would like to see.

I also am presently using a Behringer DCX2496 for crossover duties to my subwoofer. I would love to have MC perform the same functions as per Mojave's post above. My ultimate aim is to have all pre/pro duties done on the PC - hopefully by MC. I think there are lots of others that would really appreciate this ability too.

Thanks,
Rod

PS Thanks for all the good progress you have made in 14!

[Matt]

I read that the Computer Audiophile website will be doing a review of MC14 soon. Their focus is primarily on two channel audio with external DAC's and no subwoofers. However, I think it would help differentiate MC from other media players if it also had advanced subwoofer crossover capability. I have an earlier post in this thread detailing what I would like to see.

I read your old posts again, and there are a lot of suggestions.

One suggestion was to have a cross-over per channel in 'Room Correction', which is a good idea.

Are you also looking for a 2.1 output format in DSP Studio > Output Format?

What else is important?

Thanks.

[BradC]

Re crossover for subwoofer, would it be possible to route the sub channel to a different sound card (or zone). The reason is that I have in the PC a Lynx L22, which is only 2 channel, but high quality. Therefore, I would prefer to use the Lynx, but it doesn't allow the use the of sub with driver level bass management.

I would also like to add another vote for native convolution room correction in MC.

[Matt]

Re crossover for subwoofer, would it be possible to route the sub channel to a different sound card (or zone). The reason is that I have in the PC a Lynx L22, which is only 2 channel, but high quality. Therefore, I would prefer to use the Lynx, but it doesn't allow the use the of sub with driver level bass management.

That makes sense in your case, but it's really tricky.

Different sound cards can have slightly different clocks and latencies.  It's a tough problem to keep them in step perfectly on the software side.

I can't think of a nice solution, save for upgrading to a card with more channels.  And I wouldn't like that solution either if I already bought a Lynx.

[mojave]

I read your old posts again, and there are a lot of suggestions.

One suggestion was to have a cross-over per channel in 'Room Correction', which is a good idea.

Are you also looking for a 2.1 output format in DSP Studio > Output Format?

What else is important?

Thanks.

If you don't have bass management in your soundcard drivers, or if these drivers are inadequate, you need bass management for both 2.1 and 5.1/7.1 or any other combo. For example, at my office at work I use a 4.1 system. I don't have room for a center speaker under my monitor. However, I can close my office door and watch part of a movie during my lunch hour. Between my office and home, I need a 2.1, 4.1, 5.1, and 7.1 setting.

Rather than having to switch between these settings, I would prefer to set a crossover for each speaker in Room Correction with a check box for "Subwoofer and Speaker" and then in DSP Studio > Output format having a check box for "Subwoofer." If this is ticked, then bass is routed to the subwoofer using the crossover settings for each speaker. To make things simpler, I would prefer one DSP with a tab for Output Format and a tab for Speaker Settings. I am using AC3 Filter to provide bass management and matrixing (especially 4.1) for me now with both music and movies at my office and at home I can use my soundcard drivers for bass management. You should look at AC3 filter and how it is laid out. I find it very intuitive.

There are many theories on bass management. These theories are often debated/discussed at AVSForum (and other forums, too). I have tried many methods and hope to try many more. The problem with most receivers is that you are limited to using only a certain method. I purchased my DCX2496 primarily so I could try different theories and methods and decide which either sounded best or measured best (not always the same thing).

With the two check boxes I mentioned above and a crossover setting you are able to get more possibilities to try without forcing the user into one method. For example, one theory/method says that in a 2.1 system to run your mains full range and use a subwoofer to supplement the mains below their natural rolloff. To use this you would check both the "Subwoofer and Speaker" box in Room Correction and you would check the Subwoofer box in Speaker Output. If your speakers roll off at 40 Hz, then you would set the crossover for the mains to 40 Hz. Because the "Subwoofers and Speaker" box is checked, the mains still get a full range signal. Because the "Subwoofer" box is checked and a crossover frequency has been selected, the subwoofer gets frequencies filtered with a low pass crossover at 40 Hz. If you wanted more advanced settings, then you could also allow the user to specify what type of crossover and its slope.

Your goal shouldn't be to determine what method is best, but rather provide options so the user can use whatever method he likes.

An easier way to do this (I think it is easier  ) is to just have a check box that says "Use AC3 Filter for post processing" or "Use FFDShow Audio Decoder for post processing" under Options > Audio. Currently AC3 Filter only supports 5.1 and I have had some problems with ffdshow working right with my settings. I would rather be able to accomplish this with just MC.
 

[mojave]

Re crossover for subwoofer, would it be possible to route the sub channel to a different sound card (or zone). The reason is that I have in the PC a Lynx L22, which is only 2 channel, but high quality. Therefore, I would prefer to use the Lynx, but it doesn't allow the use the of sub with driver level bass management.

I would also like to add another vote for native convolution room correction in MC.

You can add another L22. Per the Lynx L22 manual:

More than one Lynx L22 card can be installed in a computer for additional audio
channels. If required, all Lynx L22’s in a computer can be configured to maintain
sample accurate synchronization during digital audio recording and playback.

It looks like each card is treated as a separate device so you may still need to use two zones. You may be able to use ffdshow or AC3 Filter to send just the bass frequencies to the second zone. You could also use a VST plugin for each zone with one zone having a high pass filter and the other zone having a low pass filter.

[Matt]

Thanks for the details mojave.

How are you wiring your 2.1 and 4.1 systems?  Normally the subwoofer isn't on its own line coming out of the soundcard in 2 and 4 speaker setups.  We haven't focused on 2.1 and 4.1 wiring configurations.

The subwoofer is on its own line for 5.1 and 7.1, and in my opinion, Media Center does a really nice job in these cases.  I especially like the Subclarity filter to remove drones and leave thumps on the subwoofer line.  I'm not sure what more we would do here except for your idea to filter bass off other speakers (although I personally prefer letting the speaker's natural rolloff work)

[mojave]

Thanks for the details mojave.

How are you wiring your 2.1 and 4.1 systems?  Normally the subwoofer isn't on its own line coming out of the soundcard in 2 and 4 speaker setups.  We haven't focused on 2.1 and 4.1 wiring configurations.

The subwoofer is on its own line for 5.1 and 7.1, and in my opinion, Media Center does a really nice job in these cases.  I especially like the Subclarity filter to remove drones and leave thumps on the subwoofer line.  I'm not sure what more we would do here except for your idea to filter bass off other speakers (although I personally prefer letting the speaker's natural rolloff work)

The subwoofer is always on its own line coming out of the soundcard if you have more than a two channel soundcard (or multi-channel output on the motherboard). You should be able to output whatever combination of speakers you want using the maximum channels on your soundcards. I have 7.1 soundcards in each of my systems. My subwoofer output on the soundcard obviously goes to the subwoofer. My other channels go directly to my 7 channel amplifier or to two 2-channel amplifiers (at the office). There are no wiring changes. If I use AC3 Filter and select 2/0+SW or 2/2+SW and select bass redirection with a crossover in its mixer then the bass is routed to the subwoofer channel. It is just like using a receiver with its crossover settings and large and small settings for the speakers. I can do this with AC3 Filter, ffdshow, Theatertek with advanced audio, WINDVD, PowerDVD, ArcSoft TotalMedia Theatre or any other software player with built in DD or DTS decoders. I can also do it with M-Audio, Creative, and Asus drivers although I could only do 4.1 with the M-Audio drivers.

Here is a quote from Arcsoft:

Having obtained DTS-HD certification, ArcSoft TotalMedia Theatre now not only has the ability to decode the lossless DTS-HD Master Audio – a “bit-for-bit” identical recreation of the original studio master sound track, DTS-HD High Resolution Audio, etc.; it also provides end users the ability to control and direct bass signals to the most appropriate speakers or subwoofer(s) in the system, a process known as Bass Management System, an important recognition from DTS. Also noteworthy are “Speaker Remapping” which offers support for various speaker playback configurations

There are two sources for the bass. One is the LFE track on the Blu-ray or DVD. This is all that is sent to your subwoofer if you don't use bass management. The other is redirected bass from the other 7 channels. The bass for the other 7 channels on DVD or Blu-ray can be as lower than 20Hz and with peaks up to 105 dB (I think the lowest bass I've seen in the 7 channels is about 8 Hz). It is very rare that the speakers (either mains, center or surrounds) can output the same dynamics and low frequency range as a subwoofer. I would bet that 99% of all home theater calibrators would use a crossover even on the mains for movies. My mains are towers and while their -3dB point is 40 Hz my dual 15" subwoofers do much better from 100 Hz and down.

What exactly does the Subclarity filter do? I have never had thumps or anything else on the LFE channel or on redirected bass to the subwoofer that wasn't intentional. If it is mastered properly why would there be anomalies? Are you sure you aren't just filtering out frequencies that your sub can't play properly?

[Pjotr]

Hi Matt,

Just my 2 cents.

I like the idea of a convolver for room correction, but to my experience so far with the free VST  “Convolver” plug-in  and REW it is only useful below the Schroeder frequency and as such only for subwoofers.

For high-end audio a 2.1 stereo system with dedicated bass-management would be already a big advantage. I really like the idea of taming room modes in the bass region which is pretty difficult in normal living rooms. For bass-management that requires multi position impulse measurements for single/multi-sub set ups. But not only that, a CB sub can gain a lot from a good shelving filter that is a neat bi-quad IIR filter, to shift the Qb and Fb of the sub neatly instead of the wacky boosting filters usually found inside subs.

REW is free and already there, probably that can serve as a measuring front-end to obtain the FIR correction files for a convolver implementation in MC?

One problem with the free VST convolver plug-in I encountered is that the sample rate of filter file must be the same as the music processed. Otherwise it gives really weird results. This must be proper handled when we deal with various sample rates of the music as with MC.

[gappie]

Mr. Mojave, i enjoy your posts a lot. very informative and even i understand them. 
and it all make sence. and maybe it would make it even easier for my unothodox setup..

so thanks
 
gab

[mojave]

Thanks, Gappie.

Now that I can use a 2.1 mode with MC, I enabled the Room Correction DSP and set the distances for speakers and sub. It was mentioned earlier that the volume levels would be changed based on distances. I confirmed that when I change the subwoofer distance setting, the volume level also changes.

When setting the correct phase between a subwoofer and the mains, the distance setting is adjusted until the volume at the crossover frequency is the loudest. I have found that this distance is typically closer than the actual measured distance, but it varies by subwoofer model. If the volume keeps changing as one adjusts the distance setting, how are we supposed to use measuring software or an SPL Meter to find when the phase is correct? It could be argued that the change in volume will be inconsequential, but if I would rather not have to worry about it. I don't mind the volume changes for all other channels but I think they are rather useless. One still has to measure the SPL and set the right volume for each channel. Can I at least get the sub level to remain constant? Thanks.

[Pjotr]

My wish too Level changes for a sub are of no use. For 2.1 stereo I don’t see any use for level changes for the L and R channels too.

What does the distance setting actually do? Apart from the level changes, does it simply adjust the delay and what is the reference then, just the seating position? Maybe for a 2.1 system a delay setting for each channel in just ms can be a handy option. Only for the sub channel, or only for the set of LR channels is a need for delay then.

Also I like the option to have a customable HP filter for the LR satellites. This can give better integration with the sub. But if once extensive true bass management becomes reality in MC all will change anyway I suppose.

Oh, and please can you make the distance setting also available in metric units. Rulers in feet are rare in Europe… (except for GB ).

[Matt]

No distance-based attenuation is applied to the subwoofer channel.

But after more consideration and testing, we believe that distance based attenuation should be applied to matched sets (L/R, LS/RS, RR/LR) based on their relative distance to each other instead of relative distance across all speakers.

This change will be in a future build of the program.

[mojave]

You are technically correct that no distance based attenuation is applied to the subwoofer channel. However, increasing the distance in the subwoofer channel compared to the mains attenuates the mains. Decreasing the subwoofer distance in relation to the mains does not cause a change in the mains volume. Is this by design?

[Matt]

You are technically correct that no distance based attenuation is applied to the subwoofer channel. However, increasing the distance in the subwoofer channel compared to the mains attenuates the mains. Decreasing the subwoofer distance in relation to the mains does not cause a change in the mains volume. Is this by design?

Like I mentioned above, a coming build revises this so attenuation is based only on matched channel sets.  Phasing still operates based on all the speakers in play, so moving the subwoofer distance could change  phasing on other speakers to provide a perfectly phased image at your listening position.

[udailey]

I would love to see room correction with a calibrated usb mic.
Most things this would solve are obvious. One that might not be so obvious is subwoofer timing. Matching the speed of the sub to the fullrange speaker. Full range single driver speakers and planars and ESLs are very hard to match with subs because the full range driver is much faster than the sub. What MC could do is to have the computer 'listen' to both sub and full ranger near the crossover frequency and slow the full ranger to flow better with the subwoofer. This would allow much better sub integration into the sound. As it is now very fast speakers are fast because they have very little mass to move. Because they have little mass to move they usually dont create low bass with much SPL so they need a sub but the sub has more mass to move and is slower. When this is the case we can hear the separate speakers. If we could slow down the full ranger near the crossover frequency then the sub would integrate much more seamlessly and we would again hear ONE speaker rather than the 2 or more that are involved for each channel.
Second point.. some subs have a servo correction system in them. Some of these servo correction systems are implemented with a mic rather than other sensors or circuits that guess. MC could act as a subwoofer servo if that mic was continuously in use. Maybe that last one is crazy but it sure would be nice.
Uriah

[Matt]

mic was continuously in use

Good idea.

This would also allow for nice volume-based equalization to adjust for the human hearing curve:
http://en.wikipedia.org/wiki/Equal-loudness_contour