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[mwheelerk]

From a hearing test today it has become apparent I have suffered some long term damage due to extend periods of exposure to loud sounds. Though I no longer play my music as loud as I use to nor to I attend live shows anymore the years have caught up to me.

I'm comfortable now in the general use of JRiver but not with some of the special features like DSP for EQ. I thought that use of this might help me overcome some of the difficulty I'm having in the 4k to 6 k band. Could someone provide some guidance and recommendations as to how I can set up utilize this type of feature?

Thank you

[mwillems]

From a hearing test today it has become apparent I have suffered some long term damage due to extend periods of exposure to loud sounds. Though I no longer play my music as loud as I use to nor to I attend live shows anymore the years have caught up to me.

I'm comfortable now in the general use of JRiver but not with some of the special features like DSP for EQ. I thought that use of this might help me overcome some of the difficulty I'm having in the 4k to 6 k band. Could someone provide some guidance and recommendations as to how I can set up utilize this type of feature?

Thank you

The wiki topic on the PEQ functions might help: http://wiki.jriver.com/index.php/Parametric_Equalizer

That said, if you can describe a little more about your frequency sensitivity , I might be able to provide more specific advice. 

For example, if you have a dip in your hearing sensitivity between 4K and 6K, I'd suggest that you go into parametric equalizer in the DSP studio and add an "adjust a frequency" filter.  I'd set the frequency to 5,000Hz, the Q to 2, and the gain to 3dB (just to start).  Then you could play with those parameters as needed (i.e. more gain or less, etc.).  Raising the Q makes the range of frequencies affected smaller, lowering the Q makes the range of frequencies affected larger.

But if you know the specific frequencies where you need to compensate, I'd be happy to help you develop a set of compensation filters.

[mwheelerk]

Thank you so much. I have tried your initial suggestion and just started playback.

The best I can say the best I can decipher this report they gave me is in the range tested I am fairly flat at 20dB (which the scale identifies as Hearing Level dB)from about 250hz to 2500hz it goes to 25dB at 3000hz (all of this falls within a range identified as Normal. It then takes a drop to 40dB at 3000hz which is right on the line shown as Mild And Moderate. From there it rises back up to 30dB and stays flat at that number to 8000hz which are in the Mild range.

The ENT doctor told me that loss in the 4-6khz range is typical from long term exposure to loud sounds. I can only assume that 50 years of rock and concerts has caught up to me. If this is a practical way for me to compensate and continue to enjoy my music. I have listen to five songs since beginning to type this reply and I do feel the boost has increased detail that I have felt somewhat lacking (maybe this is wishful thinking at this point).

Any suggestions or advice are genuinely appreciated.

Why the Parametric equalizer adjustment rather that the "standard" equalizer adjustment?

[JimH]

There is a topic around here about Matt's dad and Matt's selection of headphones for him.  He has some hearing loss.  It might have something useful.  I think it was about a year ago.  Search for "all us kids" [screaming] or [yelling].

Here it is:
http://yabb.jriver.com/interact/index.php?topic=81623.0

[mwillems]

Thank you so much. I have tried your initial suggestion and just started playback.

The best I can say the best I can decipher this report they gave me is in the range tested I am fairly flat at 20dB (which the scale identifies as Hearing Level dB)from about 250hz to 2500hz it goes to 25dB at 3000hz (all of this falls within a range identified as Normal. It then takes a drop to 40dB at 3000hz which is right on the line shown as Mild And Moderate. From there it rises back up to 30dB and stays flat at that number to 8000hz which are in the Mild range.

The ENT doctor told me that loss in the 4-6khz range is typical from long term exposure to loud sounds. I can only assume that 50 years of rock and concerts has caught up to me. If this is a practical way for me to compensate and continue to enjoy my music. I have listen to five songs since beginning to type this reply and I do feel the boost has increased detail that I have felt somewhat lacking (maybe this is wishful thinking at this point).

It sounds like your high frequency hearing (at and above 8 KHz) is less sensitive than your low frequency hearing, but that you have an especially deep dip in the 4 to 6 KHz range?  Does that sound right?  If so, you need a parametric boost between 4KHz and 6KHz (which I explained how to do above) and then a shelf filter to make all the high frequencies a little louder.

So I'd suggest a filter like the one I described above, but you might want to try playing with lowering the Q on the filter I previously suggested (try a Q of 1.5) and maybe raise the boost a little. Then, you might also want to include a "high shelf filter" with the frequency set to 5,000, the Q set to 1, and the gain set to 2 or 3dB.  That will make all frequencies above around 4KHz somewhat louder.

Once you've figured out what your final boost settings are, you should include an "adjust the volume" filter at the end turning down the whole channel by however much boost you added (i.e. if you have a +3dB shelf and a +3dB frequency adjustment, you'd want an "adjust the volume" of -6dB).  That's for safety to prevent the boost from causing JRiver to clip if you turn the software volume all the way up.  If you use internal volume in JRiver and don't usually keep it maxed, you might be not need to do the offset.

Why the Parametric equalizer adjustment rather that the "standard" equalizer adjustment?

Because it's much more adjustable; you can specify more exactly which frequencies you're trying to target, and you can do neat tricks like the shelving filter I mention above.  It's easier to make something tailored to your specific situation using PEQ, and once you've got the general idea, it's easier for you to tweak it until it's just "right."

Let me know if this helps!

[mwheelerk]

Thank you mwillems this is very helpful.

The first filter I set up with frequency of 4000 Hz, just reduced the Q to 1.5 and have the Gain at 4.5.   For the High Self I set it at 5000 Hz and Q at 1 as suggested. I set the Gain at 2.5.  I will listen for a day or so to more familiarize myself with the sound and then implement your final suggestion.

Thanks

[mwheelerk]

At this point maybe just wishful thinking on my part but it does seem my music has become more alive again for me with these minor adjustments.

[mwillems]

At this point maybe just wishful thinking on my part but it does seem my music has become more alive again for me with these minor adjustments.

I'm glad you're enjoying it.  You've made real changes to the tonal balance that are tailored to your hearing loss, so it's probably not just wishful thinking.  You can try a simple test: have someone else turn the PEQ module on and off while you listen to music (without looking at them or the screen).  If it still sounds better to you then, you'll know it's real

[mwheelerk]

I was just speaking to a friend tonight and ask him to come over sometime this week to listen with me. I'm suspecting it might sound a little "boosted" to him but it continues to sound better to me through a long day of listening. There is just a little more emphasis in the detail but it also sounds as if the soundstage slightly expanded.

[qveda]

Hi, newbie here...

just downloaded MC for mac, since I don't like the user interface of Amarra.  Not sure if there is a significant difference in audio quality, but look forward to comparing.   like others in this thread I have the common hearing loss around 4K - but mostly in the left ear.  On the desktop it won't be as noticeable, but if listening with headphones or IEM's,  I'd like to boost 4K just for my left hear. 
Question 1:  on the trial version, I don't see how to access DSP or EQ . must I buy the full version to try it?
Question 2:   can I adjust EQ for left/right separately?

[Matt]

Question 1:  on the trial version, I don't see how to access DSP or EQ . must I buy the full version to try it?

Look at DSP Studio.

Question 2:   can I adjust EQ for left/right separately?

Use Parametric Equalizer for per channel correction.

[qveda]

this may seem lame, maybe I need DSP for my eyes, too,  but where do I find DSP Studio? I don't see it on the JRiver site.

[6233638]

It's accessible via the Player menu (near the bottom of the list) or by clicking on the toolbar icon on the right with three vertical bars. (looks like an old analog EQ)

[Bluemoon]

I'm following this thread with interest. Thanks for the explanations, it's helping me as well.

For me one difficulty with the JRiver's implementation of the PEQs is the lack of visualisation of the overall FR curve when PEQs are added/removed and enabled/disabled. Having a resulting FR curve would assist a lot comparing one's own measured FR curve to JRiver's and make the necessary adjustments visually without trial & error.

[qveda]

Found it.  thanks!   next I'll try to find how to have the DSP display remain open so I can toggle, etc more easily, and I agree with the previous post

[mwheelerk]

I am continuing to utilize the tool and with minor adjustments, trial and error, I am very satisfied with the results.

[sdawson]

Hi mwillems. I'm hoping you might be able to help me with a starting point for PEQ for my particular hearing loss pattern. I've attached the relevant part of my hearing test. Thanks for any pointers!

Is there any hope that MC will ever allow these sort of plugins on streaming output? It strikes me that it could be possible during a transcode to PCM or something similar.

[mwillems]

Hi mwillems. I'm hoping you might be able to help me with a starting point for PEQ for my particular hearing loss pattern. I've attached the relevant part of my hearing test. Thanks for any pointers!

It looks like your left and right ear are a little bit different so I'm going to offer a "blended" suggestion to try and split the difference.  I'd suggest starting with the following filter shapes:

1. A high shelf with the frequency set to 2000Hz, the Q set to .8, and the gain at +3dB.
2. Next, a high shelf with the frequency set to 3000Hz, a Q of 1, and the gain on this one at +5dB.
3. Then, a high shelf at 3500Hz with a Q of 1, and a gain of +3dB
4. Finally, an "adjust the volume" filter with a -11dB setting.

Once you've dialed those in, you should put on some familiar music and tinker with the settings to suit your taste.  The first step I'd suggest is increasing or decreasing the gain of the filters (but leaving the Q and frequencies alone just at first).  If you decide you want to branch out into additional tinkering, here's an explanation of the filters and parameters to help you in tailoring them:

A high shelf with a positive gain looks something like the blue trace in this image: http://en.wikipedia.org/wiki/File:Shelving-eq.svg
You can see that the shelf in the illustration increases the volume of frequencies above a certain point, and it rises for a while and then stops.

The frequency setting determines where the center of the rise is.  The high shelf in the link probably has a center frequency of around 4,000Hz, give or take.

Q determines how quickly the shelf reaches its top (higher Q is faster, lower Q is slower).  The shelf in the link has a Q of around .5. FYI JRiver doesn't support shelves with steeper Q's than 1 (higher values can be entered, but are ignored).  That's one reason I recommended several shelving filters (more on this below).  

The gain setting determines how much rise the shelf will have total.  The shelf in the linked picture has a gain of 9dB.  

With that said, you can probably see why I recommended a few shelving filters. Your chart at the macro level looks kind of like a high shelf filter in reverse (or a high shelf with negative gain).  I recommended staggering a few shelves because your chart shows a gradual dip followed by a steeper slope; staggering a few shelves creates a steepening effect.

The total gain the shelves apply is about 11dB, which is less than the total attenuation in your chart (between -15dB and -20dB from the baseline depending on the ear).  I suggested starting with less gain because my experience has been that its best to start with less boost than you think you might need. You can always add more (or less if need be).

One important note on the 4th suggested filter: it's there to offset the total gain added by the shelves.  That step is important, or you might wind up driving your audio output into clipping.  If you're trying out different settings and increase the total gain of the shelves, you need to lower the volume to offset the new gain.  

Hopefully that will get you started in the right direction.  Please let me know if that helps at all, or if you have any questions.

Is there any hope that MC will ever allow these sort of plugins on streaming output? It strikes me that it could be possible during a transcode to PCM or something similar.

There's been talk of applying DSP to DLNA output, but the general consensus so far has been "not soon."  

In the current toolset, if you're the only one whose going to be listening to the media you can potentially use the convert format or handheld sync tools to "bake in" EQ, but if you decide to go down that road be sure to keep a back up of the original file!  

[sdawson]

This all gets confusing very quickly!

A high shelf with a positive gain looks something like the blue trace in this image: http://en.wikipedia.org/wiki/File:Shelving-eq.svg
You can see that the shelf in the illustration increases the volume of frequencies above a certain point, and it rises for a while and then stops.

The image you refer to comes from the article at https://en.wikipedia.org/wiki/Equalization_(audio). It calls that a first order response, boosting all frequencies above a certain level. There is no Q value in these types of filters. The other image in the article below this one describes a second order response, where the Q value creates a bell curve. Which one of these really describes MC's high shelf filter? Strikes me that MC's straight "adjust a frequency" filter produces a bell curve. Or maybe MC's high shelf is really like the first order response curve, but with the addition that Q can adjust the shape of the curve somewhat. I think I'm sensing a difference in terminology between the wikipedia article and MC's version at http://wiki.jriver.com/index.php/Parametric_Equalizer.

Let's take your first suggested filter. Will that start boosting frequencies from 2000Hz all the way up to 20000Hz or whatever? Then will the second one amplify that gain even more, starting at 3000Hz? How did you decide on the Q values? I'm finding it very hard to visualise exactly what Q does to the shape of the curve.

So far,  the suggested settings do make audio seem excessively bright, but I'm hearing stuff I wasn't before! Just need to try to decide what to change to strike a balance.

Your help is very much appreciated...

[mwillems]

This all gets confusing very quickly!

The image you refer to comes from the article at https://en.wikipedia.org/wiki/Equalization_(audio). It calls that a first order response, boosting all frequencies above a certain level. There is no Q value in these types of filters.

A first order shelving filter has a Q of .5

Increasing that Q will increase the steepness.

The other image in the article below this one describes a second order response, where the Q value creates a bell curve. Which one of these really describes MC's high shelf filter? Strikes me that MC's straight "adjust a frequency" filter produces a bell curve. Or maybe MC's high shelf is really like the first order response curve, but with the addition that Q can adjust the shape of the curve somewhat. I think I'm sensing a difference in terminology between the wikipedia article and MC's version at http://wiki.jriver.com/index.php/Parametric_Equalizer.

The second order filter shown on the wiki is not a shelving filter, it's a conventional parametric filter (which JRiver calls an "adjust a frequency filter").  That wiki page is very confusingly laid out (that's why I linked to the graph and not the article).  A second order shelf looks a lot like  afirst order shelf, just steeper.  Likewise, increasing the Q of an MC shelving filter maintains the same overall shape, it just gets steeper.  

There are a few terminology differences in JRiver; the term "parametric equalizer" generally refers only to what JRiver calls the "adjust a frequency" filter, but JRiver's entire family of filters is included in a module called "parametric equalizer."  Don't let that throw you.

Also it's very common to talk about "orders" of shelving filters, which can be misleading.  Q serves the same purpose here, it's just more customizable.

Let's take your first suggested filter. Will that start boosting frequencies from 2000Hz all the way up to 20000Hz or whatever? Then will the second one amplify that gain even more, starting at 3000Hz? How did you decide on the Q values? I'm finding it very hard to visualise exactly what Q does to the shape of the curve.

The first filter will start boosting frequencies somewhere around 1200 Hz, and reach the top of it's rise somewhere in the 3.4KHz range.  2000Hz is the center of the rise.  For a 3dB shelf, 2000Hz is the point where it's up 1.5 dB.  The shelf will not boost more after it reaches the top of its rise.  That is, if you input a gain of 3 dB, once the rise reaches +3dB, no new boost is added by that shelf after that point.

The multiple filters are to create a steeper and less uniform slope.  Your chart showed a gradual dip followed by a steeper dip, so I was trying to model that using staggered shelves.  I picked the Q by trying a few values that looked likely to be close, and then plugged in a few values using modeling software until the resulting slope had a shape more or less like the inverse of your hearing test.

Check out the following post, it shows two low shelves with the same frequency and gain settings, one with a Q of .5 and one with a Q of 1.0: http://yabb.jriver.com/interact/index.php?topic=86791.msg594451#msg594451 .  Those shelves are low shelves not high shelves so they rise as they get lower, but the shape is the same.

So far,  the suggested settings do make audio seem excessively bright, but I'm hearing stuff I wasn't before! Just need to try to decide what to change to strike a balance.

I was worried that the settings might make things a little bright, which is why, as I noted above, I didn't even try to add in the full 20dB of boost that your chart would have suggested. A little high frequency can go a long way!

I'd recommend lowering the gain on each of the filters a little (a dB or two each) until it no longer sounds "bright" as a first step.

Your help is very much appreciated...

It's my pleasure, hopefully we can get it dialed in where it sounds more natural to you.

[sdawson]

OK, might be getting the hang of this (I think)! Few more questions though, I'm afraid...

I agree that the wikipedia article is laid out in a very confusing way - hinders more than helps sometimes! The forum post you linked to was very helpful in visualising the effect Q has, especially the one that shows HS traces (albeit it has a negative gain), where .5 has more of a straight line than 1.

For a given frequency you set in an HS filter, how can you tell the frequency range it will actually act on? You said the 2000Hz filter would start at 1200 and end at 3.4KHz. Is there some formula to work this out? I suppose it depends on the gain you specify - is that right?

Your suggested filters do seem to overlap in frequency ranges. Do they then work cumulatively, such that the second one might amplify what the first one has done, at least towards the bottom of the second one's frequency range?

At the beginning of this post, you were recommending to mwheelerk "a high shelf filter with the frequency set to 5,000, the Q set to 1, and the gain set to 2 or 3dB in order to make all frequencies above around 4KHz somewhat louder." Would this really make ALL frequencies above 4KHz louder? It would reach the top of the shelf somewhere wouldn't it?

What does a typical treble control on an amplify actually do? I ask this because I do have one system that I won't be driving using MC, so it would be good to know what its tone control might achieve (it's a Meridian system, so its treble is controlled digitally, but I expect its effect will be the same as a standard analog control).

I'm pretty excited by the prospect of boosting my systems. I must say I've been disappointed by the performance of them lately, and they're not exactly low-end (one is Krell/Focal, the other tube/Focal). It does mean i'll have to rethink my process of going to music streamed across my network to Oppo devices, which is nice and convenient, but if equalising does what I hope it will I'll have to use laptops/Mac Minis and MC everywhere. Multiple licences here I come!

Thanks yet again.

[mwillems]

For a given frequency you set in an HS filter, how can you tell the frequency range it will actually act on? You said the 2000Hz filter would start at 1200 and end at 3.4KHz. Is there some formula to work this out? I suppose it depends on the gain you specify - is that right?

There is a formula, it's actually in that wiki article under "Shelving filter." But I'll confess I did not apply the formula, I just sort of plugged in values and tested the result in modelling software.  You're right that the range affected will vary somewhat based on the gain, and on what your definition of "affected" is (the range I mentioned above is where the rise starts in earnest and begins seriously slowing down, the transition takes a little longer to completely finish). You can look at the traces in the other thread I linked above tog get a sense of how long the rise will take

 

Your suggested filters do seem to overlap in frequency ranges. Do they then work cumulatively, such that the second one might amplify what the first one has done, at least towards the bottom of the second one's frequency range?

They definitely amplify each other, to better match your hearing chart, which shows a slow dip followed by a faster dip.

At the beginning of this post, you were recommending to mwheelerk "a high shelf filter with the frequency set to 5,000, the Q set to 1, and the gain set to 2 or 3dB in order to make all frequencies above around 4KHz somewhat louder." Would this really make ALL frequencies above 4KHz louder? It would reach the top of the shelf somewhere wouldn't it?

Think of a high shelf filter as having having three locations: the frequencies below the transition band (frequencies unaffected by the shelf), frequencies in the transition band (where the shelf begins rising), and frequencies above the transition band (frequencies above where the shelf stops rising).

A high shelf with positive gain increases the volume of all frequencies above the start of the transition band.  The gain increases as frequency increases across the transition band. Then the gain stops increasing at the end of the transition band, but frequencies above the transition band are louder than the frequencies below the transition band by the amount of "gain" set in your high shelf.  Does that make sense?

Maybe a thought experiment: imagine an "ideal" shelf with an infinite Q and a 5dB gain.  The transition band for such a shelf would be zero HZ wide, it would basically look like a single stair step.  If such a shelf were set at 4000Hz, every frequency above 4000Hz would be exactly 5dB louder than every frequency below 4000Hz.

Real shelves have non-infinite Q, so they obviously have a non-zero transition band.  The range I gave you above is a rough approximation of the transition band of the first shelf.  The real band is somewhat wider, but the bulk of the transition happens in the range I quoted.

What does a typical treble control on an amplify actually do? I ask this because I do have one system that I won't be driving using MC, so it would be good to know what its tone control might achieve (it's a Meridian system, so its treble is controlled digitally, but I expect its effect will be the same as a standard analog control).

A normal amplifier tone control is essentially a high shelf filter with adjustable gain.  The limitation is that typically the "frequency" and "Q" of that filter are fixed, only the gain is adjustable, although I have seen tone controls that also have adjustable frequencies.  Check out the graphs on the first page of this paper: http://www.sevenwoodsaudio.com/AN12.pdf

It shows both a conventional fixed frequency tone control and a tone control that allows for sweeping the frequency.

I'm pretty excited by the prospect of boosting my systems. I must say I've been disappointed by the performance of them lately, and they're not exactly low-end (one is Krell/Focal, the other tube/Focal). It does mean i'll have to rethink my process of going to music streamed across my network to Oppo devices, which is nice and convenient, but if equalising does what I hope it will I'll have to use laptops/Mac Minis and MC everywhere. Multiple licences here I come!

Thanks yet again.

Sounds like you're getting there.  As for computers everywhere, welcome to my world.  There's a lot of flexibility in a system like that.  A JRiver client on a computer offers a lot of advantages over a more conventional access point.

I've said it before, but JRiver basically changed everything about the way I listen to music except for the chair and the guy in it 

[sdawson]

Brilliant stuff. Very confident I've got it now, thanks to your amazing help!

Currently listening to music using your filters, but each reduced by 2db, and I must say it sounds pretty good.  The full dB you suggested is certainly WAY over the top for some reason. Lots more playing to do of course!

I guess there's no way of telling what frequency range a particular treble control implements, but it may have to do on one system...

I've been so fixated on "bit perfect playback" for so long - I've probably missed out on so much because of this!

[mwillems]

Brilliant stuff. Very confident I've got it now, thanks to your amazing help!

Currently listening to music using your filters, but each reduced by 2db, and I must say it sounds pretty good.  The full dB you suggested is certainly WAY over the top for some reason. Lots more playing to do of course!

You might be running into one of the paradoxes of human hearing loss.  Typically when hearing loss occurs, the threshold of audibility increases, meaning you need a louder sound to hear anything at step one, which is what one would expect.  The counterintuitive part is that simultaneously the threshold of pain decreases, meaning that the volume it takes to make you uncomfortable will be quieter than it used to be.  The practical effect is that the listening volume "sweet spot" where things can be heard but aren't uncomfortable typically gets narrower with hearing loss.

That's one of the reasons I recommended playing with the volume on the filters first; you want just enough, but not too much    

It's also possible that the dB scale on your hearing chart isn't perfectly calibrated, or that the conditions of the test are different enough from music that the absolute scale isn't the same.  

Regardless, it sounds like you're on the right track!

I guess there's no way of telling what frequency range a particular treble control implements, but it may have to do on one system...

Sometimes it will be in the manual.  One potential workaround if you can't find data: download a free RTA (Real Time Analyzer) app for your phone, play pink noise on the system (JRiver includes a clip) and turn the tone control all the way up.  The phone RTA apps are not very sophisticated, and I wouldn't use one to do serious calibration, but they're usually good enough to identify the region the tone control is working in.

I've been so fixated on "bit perfect playback" for so long - I've probably missed out on so much because of this!

I'm a big believer in neutral sound reproduction, but I think bitperfect playback is overrated as a way of achieving that. If your goal is to hear the music "as it was mastered," sometimes DSP can help with that, not hurt.  

One way to think of it: speaker designers correct dips and peaks in a speaker's frequency response either using electronic filters as part of the crossover design for the speaker, or using DSP in some powered speakers. Their goal is to get a neutral output from a neutral input.  Speakers are imperfect transducers and electronic filters and/or DSP allow designers to make them "more perfect."

But our ears are imperfect transducers too.  I think most people wouldn't balk at the idea of using DSP to correct a measurable non-linearity in the response of a loudspeaker transducer (i.e. rarely is it suggested that audio purity is compromised by a notch filter in a speaker crossover). So why is correcting measurable non-linearities in our organic transducers any different?  

I'll be interested to see where you land for your final correction  

[mwheelerk]

I have continued to use the initial settings with slight tweeks but based on the continued conversation I may try to incorporate a little broader scope based on the test results I received.

[sdawson]

You might be running into one of the paradoxes of human hearing loss.

That's very interesting stuff.

It's also possible that the dB scale on your hearing chart isn't perfectly calibrated, or that the conditions of the test are different enough from music that the absolute scale isn't the same.

Yes, entirely possible. I have a call in to my audiologist, and I'm expecting a call back tomorrow. Apparently he's a musician and sound recording guy, so he might be able to shed a lot of light. Here's hoping!

So why is correcting measurable non-linearities in our organic transducers any different? 

Wonderful stuff! And if bit perfect is so good, how come every speaker sounds different, so we end up just picking one that sounds good to us anyway? To one person that speaker might sound neutral and far off that to another. So I'm now very happy (with your help) to use DSP to achieve a sound that is good for me. I've down a few experiments with pink noise from JRiver using an RTA app on my iPhone, and uploaded some screen shots.

No PEQ is from USB to a DAC straight into integrated Krell amp.
PEQ high is the same with your initial suggested settings.
PEW low is the same with those settings each reduced by 2dB
Treble flat is from optical to a Marantz surround processor (so I can test its treble control), then into Krell.
Treble +5dB is the same with treble turned up to +5.

My observations:
I don't know why the high frequencies always drop off suddenly. Maybe a function of the pink noise being generated?

Your settings seem to be doing their job, for sure, although can't tell how far they go because of that drop off. My reduced ones just take a lot of the obtrusive (to my ears) edge off.

The treble control seems to do a somewhat similar job, and does, in fact, sound pretty similar. This is good because it means I can put less important sources, like Rdio, through the Marantz and still get something decent.

Think of anything to add?

[mwillems]

My observations:
I don't know why the high frequencies always drop off suddenly. Maybe a function of the pink noise being generated?

The roll off is (I would guess) primarily a result of the limitations of the microphone in your phone.  Smart phone microphones are much better than they used to be, but they're still not perfect by any means.  They're optimized to provide a good neutral reading in the voice band, but anything at either extreme is likely to be skewed.  My experience has been that you can't really trust phone readings above about 8KHz or below about 60Hz as they become drastically less sensitive in those regions.  That's part of why I said above that I wouldn't suggest using a phone RTA to do serious calibration, but it's good enough to give you the general idea (once you know to discount the edges).

The other potential "contributing factors" to the HF roll off are your speakers (many commercial speakers roll off between 10K and 20K) or your measurement position (HF waves are very directional, so, for example, if the mic were not on axis with the tweeter it might show less response between 10K and 20K).

Unfortunately, if you really want accurate readings at the top and bottom of the range, you'll need a calibrated microphone (which are not cheap), and RTA software on your computer. The cheapest calibrated mic that's self-contained (i.e. doesn't need extra equipment to work) that I know of is a USB model (UMM-6) at parts express for $60 http://www.parts-express.com/dayton-audio-umm-6-usb-measurement-microphone--390-808

If you do decide to pick up a microphone and have an interest in doing more extensive measurements, I wrote a how to on speaker measurement and correction over in another part of the forum: http://yabb.jriver.com/interact/index.php?topic=87538.0

Your settings seem to be doing their job, for sure, although can't tell how far they go because of that drop off. My reduced ones just take a lot of the obtrusive (to my ears) edge off.

The treble control seems to do a somewhat similar job, and does, in fact, sound pretty similar. This is good because it means I can put less important sources, like Rdio, through the Marantz and still get something decent.

I agree that it looks like the treble control starts working in a similar region, which is good news.  The slope isn't the same, but it sounds like it sounds close enough that it will probably work just fine.

Think of anything to add?

Not off the top of my head, and it looks like you're moving in a good direction!

[sdawson]

Spoke to my audiologist. Didn't have much more to add, other than hearing tests are measured in dBHL, which are somehow different to normal dB's. He also said that a 5dB ramp should be more than adequate. So on the right track.

Couple of other issues:

When using USB to a DAC, is there a definitive answer to which is the best type of volume control to use? I guess when using a DAC I tend to keep the volume control at max, if that makes a difference.

Secondly, since I need to have the "adjust the volume" filter, the volume between the computer source and other sources is mismatched. Is there a way to overcome this in JRiver?

Thank you once again for your amazing help.

[Matt]

Secondly, since I need to have the "adjust the volume" filter, the volume between the computer source and other sources is mismatched. Is there a way to overcome this in JRiver?

Not really.  You'll output at a lower level from JRiver if adjust the volume is enabled.  Other devices will output at full level.

You can use "Peak level normalize" from DSP Studio > Adaptive Volume to raise the level (but still maintain volume leveling).

[mwillems]

Spoke to my audiologist. Didn't have much more to add, other than hearing tests are measured in dBHL, which are somehow different to normal dB's. He also said that a 5dB ramp should be more than adequate. So on the right track.

Couple of other issues:

When using USB to a DAC, is there a definitive answer to which is the best type of volume control to use? I guess when using a DAC I tend to keep the volume control at max, if that makes a difference.

I use internal volume in JRiver because it is extremely convenient and JRiver's digital volume control is top notch.  I'm not aware of an advantage to controlling the DAC's volume as opposed to using internal volume in JRiver, other than internal volume is more convenient in some circumstances.

There are reasons why you might want to control the volume at an analog step later in the chain (maximizing the digital output volume maximizes the effective signal to noise ratio of your DAC), but my opinion is that whether those gains are meaningful or audible depends a lot on your system configuration and what kind of volume control your amp has.  But any gains to be had are had from controlling volume in analog after the sound has already left the DAC.

Secondly, since I need to have the "adjust the volume" filter, the volume between the computer source and other sources is mismatched. Is there a way to overcome this in JRiver?

Thank you once again for your amazing help.

One potential workaround is to use JRiver's WASAPI loopback functionality to route other computer audio through JRiver.  That has the advantage of applying the DSP (including volume) to all audio output of the computer, but the disadvantage that you need to turn on loopback when you want to hear sounds outside of JRiver.  

Actually that "disadvantage" can be an advantage in some circumstances: for example, I have my DAC output hooked up directly to the inputs of block amps.  It's sometimes to have to "opt in" to allow system sounds through so I don't get a very loud surprise.

If that sounds like something you want to try, here are some instructions to get loopback working (adapted from a very helpful post by forum user mojave):
1.  To use the loopback you have to have another soundcard in the system. Most folks use the motherboard soundcard. Nothing is actually connected to it, but the drivers still need to be installed. You might need to turn autosense off in the motherboard drivers.
2.  Set the above soundcard to the default soundcard in the Windows Control Panel.
3.  Set JRiver's Audio output to your normal DAC output.
4.  Before playing, say, a youtube video in an external browser (or any other time that you want system sounds), put JRiver in loopback mode by going to File> Open Live> WASAPI Loopback or File > Open URL and enter "live://loopback."

[mwheelerk]

I have continued to follow the posts in this thread with interest but just got confused regarding the mention of WASAPI. Hasn't this discussion been particular to Mac (aren't we on the JRiver for Mac board)?  I have no objection but it just confused me a bit.

[mwillems]

I have continued to follow the posts in this thread with interest but just got confused regarding the mention of WASAPI. Hasn't this discussion been particular to Mac (aren't we on the JRiver for Mac board)?  I have no objection but it just confused me a bit.

Yes, sorry, I forgot we were in the Mac section; the loopback discussion does not apply to the Mac version.  Sorry for any confusion.

[sdawson]

Not really.  You'll output at a lower level from JRiver if adjust the volume is enabled.  Other devices will output at full level.

You can use "Peak level normalize" from DSP Studio > Adaptive Volume to raise the level (but still maintain volume leveling).

Thanks. Matt. I'm guessing peak level normalise will have minimal effect on a track that has peaks up in the 90's though right? I might just have to live with cranking up the volume control on my pre's/amps I guess.

I use internal volume in JRiver because it is extremely convenient and JRiver's digital volume control is top notch.  I'm not aware of an advantage to controlling the DAC's volume as opposed to using internal volume in JRiver, other than internal volume is more convenient in some circumstances.

There are reasons why you might want to control the volume at an analog step later in the chain (maximizing the digital output volume maximizes the effective signal to noise ratio of your DAC), but my opinion is that whether those gains are meaningful or audible depends a lot on your system configuration and what kind of volume control your amp has.  But any gains to be had are had from controlling volume in analog after the sound has already left the DAC.

What type of MC volume control will affect the DAC's volume control then? I have tried to read up on the various volume control options, but am still a bit confused (sorry).