Digital Recorder with frequency response down to

The product spec's for all these recorders seem to indicate a low frequency capability down to 20 Hz (some are higher, I've not seen any lower). Do you know what determines this? Is the recorder actually capable of measuring signals down closer to DC...even Hz or 0. Hz would be good.
My goal is to record vibration signals using the external MIC input, which may be in this low frequency range. So I'm wondering if the 20 Hz limit arises from microphone/speaker limitations more than the recording circuitry itself.
Appreciating any help...

What kind of digital recorder are you using? The SoundBlaster cards, for instance, are DC filtered, so that low frequencies are attenuated. But I've found that they're still measureable down to 5Hz or so, but 0.5Hz, probably not.
testsrus wrote:
The product spec's for all these recorders seem to indicate a low frequency capability down to 20 Hz (some are higher, I've not seen any lower). Do you know what determines this? Is the recorder actually capable of measuring signals down closer to DC...even Hz or 0. Hz would be good. My goal is to record vibration signals using the external MIC input, which may be in this low frequency range. So I'm wondering if the 20 Hz limit arises from microphone/speaker limitations more than the recording circuitry itself. Appreciating any help...
Message Edited by Katman on 2-5-2005 07:46 PM

Similar Messages

  • Can you use a digital recorder with Skype on a Mac mini?

    Can you use a digital recorder with Skype on a Mac mini?

    Have a look at http://www.ecamm.com/mac/callrecorder/

  • Using a portable digital recorder with MAC

    my bad - I just bought a panasonic portable digital recorder (about $70).
    I was assuming I could use it with my Mac OS X - I don't know why I assumed that, I just took it for granted that these recorders were like digital cameras, they would automatically work with any computer that had a USB input.
    wrong! when I connect it to the Mac, the recorder lights up and says "PC," as it is supposed to, and then, when I go into "about this mac" in the system profiler, it finds something on the USB port -
    but other than than that, I don't know how to access what's on the recorder - it doesn't mount on the desktop like a camera (I assumed it would).
    Can anyone help? is there any software out there that can read this little recorder - it's the Panasonic RR-US395.
    thanks!
    w
    powerMac G5 single processor   Mac OS X (10.4.1)  

    What software came with ? I have used DVRs for years. The Olympus is what I currently use. Still using an older model though as I like the cradle type computer connection. All was fine until 10.4. I got it going on the MBP finally but gave up on the G4 as I don't boot it into 10.4 anymore and it does work fine in <3.9. It was a combination of timing on the install on the MBP. Plugged the unit in about 1/2 way through installation and had to have it connected and restarted before it actually worked. Random luck. It started asking me for a "key" number and no docs mentioned anything about this so I plugged the unit in and it continued to install. Installation would hang prior to this point if the device was plugged in first though.
    Do any apps try to launch when connected? I would also check iTunes and GarageBand or any other video/audio apps that you may have to see if it's "recognized" and shows up there anywhere. Does iPhoto think a camera is connected? As by design that is/was Apple's goal with 10.4. Ultimate connectivity. Everything that connects to your machine these days is categorized by class or "family" of products. Specific products i.e. Panasonic RR-US395, are "children" This is supposed to simplify things for HW/SW manufacturers creating products that are compatible with the Mac and if they don't Mac should be able communicate with it anyway providing basic I/O functionality void of product specific features. Reading your post I realized I have not even seen a reference to DVRs as being a device "family" and I am wondering if this has something to do with it as the MBP required me to install DSS Player.
    It is my contention that there is a serious issue with this I/O Kit (actually at the kernel level that the kit communicates with) and I am curious exactly what kind of "something" Profiler sees. A screen shot of the page w/ all the lists extended would be a great help.
    There is a Dictation/Transposing SW available I have since lost it and don't remember the name. Will try to hunt it down again as it was very useful, I downloaded it used it a few times and it worked great. Found a couple others with it that were free as well. Worked with various different DVR's and formats MacUpdate maybe?
    If there is no other way to connect it you can get a cable to go from the headphone jack on the DVR to the line in on the G5 record it w/ QT, GarageBand or the like.
    If you still have 9 installed Simple text allows you to record sounds as well.
    If you can return it Olympus does include Mac SW.
    FWIW

  • How do you sync sound from a digital recorder with video?

    Can anyone tell me how to sync sound with video? Not a voice over but Voice recorded when video was shot. Voice was recorded on hand held digital recorder.

    When you're shooting use the ciak , then when in imovie, sync the sound of ciak with the closing of it and you're done
    If u havent's used , just seek for a sound or a part of speech witch is very evident to sing
    Do it before cut video
    hope helps
    J

  • Digital Recording with MacBook TOSLINK

    I am having problems making a good recording with the TOSLINK port on my MacBook. I my audio source is a dj mixer sending SPDIF through a HOSA SPDIF/TOSLINK converer into my mini-TOSLINK port I get strange fluctuations in volume and generally bad audio.
    Does anybody have luck with recording via the TOSLINK port? Could it be the bitrate coming outo f the mixer?

    I use the Canon EOS Kiss Digital N, which in the US is the Rebel XT or something. I also use an external card reader, and the MacBook works fine with the camera.
    I also use Adobe Lightroom and Photo Elements 4.0 to process and manage the photos, and they both run very well on the MacBook.
    You'll have a lot of fun, it seems like the MacBook and MacBook Pro were just made for this stuff.

  • Audio digital recording

    Can I use a audio digital recording with a window format ( it a lot cheaper ) and then transfer the recording into my computer to burn to cd or into my ipod?
    imacG5 ,iMac7,1,2 2Ghz Intel Core 2 Duo
    Austin

    Yes, .wav or ,Mp3. with USB connection. I finding the Mac compatibles are about 200.00. The PC is about 60.00. I want to record some of our tele-conference classes, instead of using a cassette recorder, which is what I do now. I would like to place the file on the desktop and burn it to cd or put into my ipod for later listening. Will imac see the recorder? Any suggestion?

  • MacBook core duo Sept. 2006  - the audio is a mystery.  Any analogue out has bass boost with bass distortion.  With digital out, by USB or Airport Express Airtunes, the frequency response is normal.  Somewhere, Apple put in an analogue bass boost.. why?!

    Since new, my Macbook core duo has sent all audio to all analogue outputs with frequency distortion.  Some physical hardware or firmware in the analogue section adds a bass boost to the frequency response of the audio....  And, this boost adds bass frequency distortion to most, if not all, of the audio analogue output.
    This happens for all audio sources, players, iTunes, videos, streaming audio/video, movies...  any sound source available to the Macbook.  No-one with whom I have talked about this problem, has ever heard of it.  Not elegant ideas for reducing it are to use equalisers on players and iTunes.  iTunes even has a preprogrammed "bass reduce" on its equaliser, as if it knows already that this inherent bass boost "feature" has been built into the Macbook, and cannot be defeated by the user.
    The digital audio is of sufficiently high quality to play well on a very good to excellent sound-system;  so, it's a shame someone has mucked around with the frequency response of the analogue conversion, by designing the frequency distortion right into the computer.  This motherboard, (which includes the sound-section), has been replaced, along with the speakers, and everything sounds exactly the same as it was before.
    To receive a flat frequency response and no frequency distortion, I listen on one receiver using Airtunes, (it doesn't matter whether or not the analogue or digital output jack is wired analogue or connected via optical cable....  the freq. response is flat);  and I listen on a high-end stereo system using an USB output port to an A/D converter, passing the analogue result using long patch-cords connected to the "line-in" jacks of the stereo.  The bummer is that most of my audio sources are not derived from iTunes...  hence, I cannot use Airtunes with Airport express for them.  I've tried using "Airfoil" without luck.  The sound becomes distorted with sibilance and other frequency anomalies.
    Question:  has anyone else discovered this sound imperfection in any Mac product?  And, does anyone know if Apple is aware of it?  Finally, has anyone found an Apple fix for the problem;  or, at least, come up with better solutions than I have?
    Thanks heaps for any impute and answers you may supply!!  junadowns

    Army wrote:
    This might not help you a lot, but if you want a stable system, try using packages from the repo wherever possible, look at the news before you update your system and don't mess things up (like bad configuration etc.).
    When it comes to performance, you won't gain much by compiling linux by yourself! Just use the linux package from [core] or if you want a bit more performance, install the ck-kernel from
    [repo-ck]
    Server = http://repo-ck.com/$arch
    (this has to go to the bottom of /etc/pacman.conf)
    (use that one which is best for your cpu (in your case this might be the package linux-ck-corex).
    Hmmm, Linux-ck-corex doesn't even load.. I am now trying to install the generic one. Hope it works.
    Edit: I will first try linux-lqx...
    Last edited by exapplegeek (2012-06-26 18:33:31)

  • Frequency response measurements with pxi-5922

    I’ am using signal express and the pxi-5922 digitizer together with the AWG pxi-5441 to analyse the frequency response of a buffer amplifier. See the attached signal express file. Many different ways to measure the frequency response have been tried and this is the best I came up with. It is basically two tone extract steps in a sweep loop. But I’ am still uncertain if this is the best way to do this kind of measurement. The fact that the detected frequency differs between the two channels worries me, even when the two channels of the pxi-5922 are looped.  Is there a more accurate way to determine the frequency response?
    Best wishes
    Stefan Johansson, SP
    Attachments:
    sweep.JPG ‏397 KB
    Frequency Sweep funkar.seproj ‏81 KB

    Claudia-
    Thanks for the response.
    Regarding the CJC- When I switch it on, the temperature readings I get are very random, roughly negative 1 degrees. (I am operating right now at room temperature, and will be using J-type TC's to measure ~43 degrees C). Also, when I use the built-in CJC, the aquisition rate seems to slow down considerably. When I use the "user specified" everything seems to be ok, including the aquisition rate.
    I measured the resitance of the Thermistor on the TBX-68T and it was about 5000 Ohms, as expected.
    Just to make sure: When using the TBX-68T, do I need to hard-wire a thermocouple to Channel 1/auto-zero and another to channel 0/CJC? Because I connected a TC to channel 0 right now, but I wasn't 100%
    sure.
    I've attached my main vi and two sub vi's that I am using for the voltage aquisition part of my project. (Note:the current measurements are just voltage measurements multiplied by the recipricol of the resistance it was measured across, ie. 10).
    I would like to keep this file as is because it writes to a file exactly the way I want it to. I'd like to have the temperature aquisition with the 4351 in the same vi as the 6030E so that they both stop and start at the same time. I am just not sure how and where to log the temperature data since there will be fewer data points than the voltage data. Any suggestions? Should I write two separate files? can I somehow append them?
    Thanks again. Hope to here from you soon.
    Attachments:
    EBlackMainDAQ.vi ‏107 KB
    Save_Data8.vi ‏45 KB
    Build_String_Array5.vi ‏33 KB

  • I have a digital voice recorder with a 3.5mm mic and headphone jack and want to transfer some recorded lecture to my mac book pro.  The mac book does not have 3.5mm  jacks.  Does anyone know if a jack to USB converter would work?

    I have a digital voice recorder with a 3.5mm microphone and headphone jack and want to transfer some recorded lectures to my mac book pro.  The mac book does not have 3.5mm  jacks.  Does anyone know if a jack to USB converter would work?

    Is there a pattern to the time of day or other detail that may be
    traced back to a OS X system cause of this odd phenomenon?
    Are there any copies of system files on any of the attached USB
    external drives? Any libraries, such as iTunes, iPhoto, etc?
    Once the drives are indexed by Spotlight, are their permissions
    ignored by the OS X? Content, if neutral, should not affect the
    wake or sleep cycle; especially if they're ignored by the OS X.
    Could there be a bad cable or other component? If so that would
    be a difficult process of elimination to detect it. Usually replacing
    most suspect components in the USB stream (external to Macs)
    is a rote way to mechanically test that idea; & not 100% sure.
    Does the equipment all have a good ground to the utility or house
    electrical field? An intermittent ground may affect more than sleep.
    Hard to say at this point. Maybe a late-night talk radio guru on
    remote viewing could peer into your situation and sleuth it out?
    Sorry to have run out of ideas, but the process must be electrical
    & mechanical to some extent. - Or perhaps odd software inspired.
    Do you have any phone-home spyware items inside, just jumping
    at the chance to spill your information? Little Snitch may help.
    PS: Perhaps the computer needs to go into Apple & have a genius
    or product specialist at AASP test the unit thoroughly... BlueTooth?
    see:
    https://www.google.com/?gws_rd=ssl#q=Wake+reason:+XHC1
    Good luck & happy computing!
    edited 2x

  • Frequency response requirements for headphones with CMSS on XFi ???

    Hi,
    I would like to know if someone could tell me what kind of heaphones are suitable for the CMSS mode with the XFi.
    I mean between : flat response/free-field correction/diffuse-field correction.
    Applying HRTF filtering should mean that headphones with flat response is the best option ( same configuration as binaural recordings).
    But I have a big doubt that Creative team expects costumers to possess such a pair of headphones, as it is rather for scientific uses (psychoacoustics, audiology etc...).
    So, if we look at the technical solutions for wide audience we have two options (FF correction and DF correction). Here is a trick because these corrections intend to reproduce some of the effects from HRTF (for two different environment configuration of HRTF measurements). It is why the frequency response of most of the headphones have a notch between 4Hz and 0 kHz.
    To simplify, if we listen binaural sounds with classical headphones the effect of outer pinna is reproduced twice.
    So I guess Creative have implemented a kind of normalization/equalization/correction process to deal with the non-flat frequency response of headphones, but do someone know if they have chosen diffuse field or free field correction ?
    This post might seem a detail but the issue can be very important for the accurate localisation and the coloration? of 3D sounds with headphones.
    Thank you, and please forgive my english!

    The only possibility that I can think of is that 2/2. mode is NOT as simple as headphone mode with crosstalk cancelation. Perhaps the HRTF only kicks in for sound sources outside of the arc directly in front of the listener. If that were the case, you wouldn't percei've any distortion for sound sources in front of you.
    Also, you are wrong regarding DirectSound3D. Keep in mind that Direct3D and DirectSound3D are not the same. The whole point of OpenAL and DirectSound3D is that they present an API to the programmer through which there is NO specification of the number of speakers. When using OpenAL or DirectSound3D, the only thing a programmer can do is specify the location of a mono sound source in 3D space relati've to the listener. The speaker settings for your DirectSound3D or OpenAL device will then determine how this sound is "rendered" by the soundcard. It is not under control of the game. For example, if you have 5. speakers and the 3D position is behind you, the SOUND CARD will make the decision to use the rear speakers. If you use headphones, the SOUND CARD will decide to apply an HRTF to create the illusion of a rear sound source. The point is that the game does not have control over how many speakers you will get sound from.
    However, to further complicate the situation, there are SOME games (HL2 is an example) where DirectSound3D is used, BUT the sound output of the game itself IS a function of the Windows speaker settings. This is not how programmers are SUPPOSED to use DirectSound3D. I've written about this countless times. There is a good post on [H]ard|Forum about this. Do an "advanced" search with my username (thomase) looking for the terms "hl2" and "cmss".

  • HT3775 can a .mov file be recovered if the record was accidentally shut down from loss of power? i was recording from the AJA Pro Ki when we lost power, the file is there but it wont play and comes up with the message, "The document "SC1ATK63.mov" could n

    Can a .mov file be recovered if the record was accidentally shut down from loss of power? i was recording from the AJA Pro Ki when we lost power, the file is there but it wont play and comes up with the message, "The document “SC1ATK63.mov” could not be opened. The movie is not in a format that QuickTime Player understands."

    Hi,
    Troubleshooting a start up script can be difficult. There are some third party programs that also keep logs of start up programs, however for Firefox this may be different.
    Is Firefox a startup program? [http://www.winxptutor.com/msinfo32.htm]
    Its also possible to check the Web developer tools for any scripts in a page: [https://developer.mozilla.org/en-US/docs/Tools/Debugger]
    In the control panel there is also Administrative tools to view event logs, but this may be something a local technician can walk you through.

  • Which digital recorder works with a Mac Mini?

    Has anybody bought a Digital Recorded that can attach ti a Mac Mini? I've read up on the Olympus and Sony ones, but it seems none of them work with a Mac ? Do any of these recorders attach to a Mac through a USB ??? directly to a Mac ???

    Olympus' DS-330 connects via the USB port, and works for me...But I believe it's a discontinued model. Olympus posts their instruction manuals online!

  • Digital recorder audio not syncing with video footage

    Am I missing the obvious and easy method? When we shoot weddings, we always use a separate digital audio recorder and lav in the groom's pocket as a backup device. When it comes time to use that audio, it always goes askew compared to the video's audio. My guess is this happens because the video is only recording 29.97 frames per second and not a full 30, whereas the digital recorder provides a real-time recording. So, I end up having to shorten the audio every 10 seconds or so (by cutting out a few frames worth of audio) to keep everything synced. Is there a way to convert the audio from the pocket recorder to a video format so it will stay in sync without the need for editing it? Thanks!

    I tried the resampling and even though original file was 44.1khz and I did convert it to 48khz, after a few minutes go by the file still goes seriously out of sync. It's a 34 minute audio clip, and by the end of the clip, its about 3-4 seconds out of sync. Any other thoughts? Thanks!

  • FS7 audio Frequency Response 50Hz - 20KHz?

    Hello guys, I'd like to point out an issue that I couldn't find on the forum regarding the audio frequency response.I've always worked with pro camcorders that recorded 20Hz to 20KHz, which is the standard frequency us humans can hear. So I was surprised when I saw in the FS7 manual that the audio Frequency Response is 50Hz to 20KHz. Is this a typo? Or is the camera actually limited to 50Hz for the lower freqencies?I would assume that the internal mic would be limited to that range, yes, but not the internal recording of the sound? Why limit the camera to 50Hz? Any light on the subject would be appreciated. Thanks!

    The tests I posted are for the mic inputs, not line inputs. I did test both, but since I often run & gun and cannot support all the extra gear to make the FS7 mic inputs on  a par with Sony's EX3 mic inputs, I'm requesting that the high pass filter added with firmware 2.0 be removed in futre updates. We have the wind filter already. No additional filter is needed. While Genelec monitors are pretty good, they're not noted for deep bass response. Most studios use the ubiquitous NS10, which are only good to 80hz, which is why few people notic the deficiency. The custom built system in my screening room has a -3dB point of 7hz. I noticed that even piano sounds thin on this camera. that doesn't surprise me,  since the rolloff starts at 200hz. All I'm asking for is the same quality I'm used to with my EX3's mic inputs. Only 0.5dB down @ 20hz and no thinning of the midrange. When I do large budget orchestra shoots (rare) I use 24/96 sampling, 8 channels and large diaphragm studio condensers. My recordings earned critical praise from Peter Aczel at The Audio Critic. I was a sound engineer for 4 decades before I got my first digital camera. In addition to that, I was consulted as an independent expert on infrasonics for chapter 17 of Ethan Winer's THE AUDIO EXPERT, published by Focal Press. I was responsible for the information on subwoofers and how they operate. Acoustic and speaker design share a long history with me, and my 'day job' is amplifier design, modification and repair. I'm too intimately aware of what's going on in the signal path of an audio system and when modern digital systems deviate intentionally from DAT-quality, it really bothers me. I made a big deal about this in 2007 with the Sony HVR-V1U, and Sony must have listened, because they got it right with the EX1 and EX3 cameras. But sadly, they've reverted to these games with the FS7, though not as badly as 2007.

  • Flat Frequency Response

    I probably shouldn't have to be asking this question since I charge people for my obviously amateurish recording abilities but it's one that I've never had explained to me and one I need to know the answer to......
    Let me set the question up this way:
    When you get in the car and pop in a professionly produced cd, most people crank the treble control up to 6-10 (on a scale of 10) and the bass up to (4-10) depending on the factory speakers and type of material and whether or not they care what their music sounds like. When you get home and you're listening to the much higher end home stereo still listening to that professionaly produced album, you still reach for those treble and bass knobs and crank them up several notches or if you have a graphic eq you tweak out a smiley face .
    There's so much emphasis in the recording world about getting a flat frequency response out of your room with absorbers and bass traps and spreading around the reflections with deflectors, etc...etc..etc..., that we spend thousands of dollars on this stuff and some measuring software to make sure that it's flat. Then we use that flat response to produce music that sounds great and expect that to translate to those cd players and home stereo systems.
    (I'll additonally preface my question by saying that I've had no problems getting my music to translate from my home studio to any other playback system, but I'm a little confused about what's going on.)
    Now finally my question(s), when we reach for those treble and bass knobs on our car and home playback systems, are we really just trying to make up for the lack of bass and top end in those systems so that we too can achieve a flat frequency response and make the music sound good on whatever system?     or
    Do we as listeners actually prefer the smiley face frequency response in music and are we taking a cd that itself has a flat frequncy response and making a smiley face out of it so that it sounds good to our ears? (Please don't give me a material/genre answer.)
    The reason I ask is because I have to put a graphic eq on my Truth 2031A monitors to make the professional stuff sound good through them, and then I in turn mix my music to sound the same for whatever material/genre of course. (I'm not really interested in any monitor bashers or I would've asked this over at Gearslutz.)
    So again rephrased...Do the masses think music sounds good when it has a flat frequency response or the smiley face and if it's the lattter of those, how are we supposed to achieve that when our home studio setup is producing a flat frequency response, do we tune our monitors with eqs like me?
    Additonaly I understand that when were talking about flat frequency responses in rooms we're talking about throwing sine waves through a system and ranging their frequencies, measuring them out so that we can detect any over emphasis/deficiencies in the room so maybe this question is a little more towards monitor tuning.

    If you look up Fletcher and Munson in Google, you might begin to get a bit of the start of an idea of why this isn't quite as straightforward as it seems.... and I'm not sure that I can give you a complete answer either, although I can give you a few connected but slightly random things to ponder, wearing my acoustician's hat:
    The fundamental problem is that when things are quieter (and less distorted, incidentally) our ears get more sensitive to the midrange frequencies, and if we listen to music that way, it invariably sounds as though the bass and treble are out of balance. In cars it's slightly different though; the frequency response of whatever's in there generally tends to be anything but flat - and often over-emphasises the midrange anyway. Treble tends to get absorbed very easily in upholstered cars, and since most car speakers don't have anything like acceptable tweeters in as a rule, it's not surprising that people want to increase the treble. As for the bass - well I'm always turning that down personally, but I know what you mean in principle!
    If by a 'commercial' CD you mean one where the vocal is prominent, then yes I can easily imagine why you might as a matter of course want to increase the response at the extremes - it makes sense if you think about it. The mid-range vocal is prominent and probably compressed, so its average level is louder than the backing - this helps it to stand out. But also it distorts the overall time-based response - the backing may well be balanced so it's okay on its own, but that doesn't always translate if you have the wrong vocal settings applied, or at a minimum, applied unsympathetically. And some voices make this significantly worse; for instance Sealion Dying (AKA Celine Dion) makes the most appalling racket in the midrange, and you'd definitely need less of that!
    So really I'd say that it's not a Bass/Treble issue, but a midrange one. If you look at commercial CDs in general, you tend to find that the energy distribution is pretty even over the whole audio band, which implies that it falls off at 6dB/octave if you look at it in Audition (this is an energy/Hz thing), but in reality most CDs these days are mixed a little brighter than that - more like a -3dB/octave slope down from about 1kHz, and that's partly to compensate for a lot of things - some of which are cars... You do have to watch out for the distortion issue though - most people don't realise, but you are able to tolerate rather higher levels of non-distorted sound than anything with significant distortion levels in it, and if that distortion is in the midrange, then you'll want it quieter anyway. So decent, over-rated under-run PA systems always sound cleaner and louder but you should beware - they can damage your ears just as much, if not more.
    Do car interiors themselves increase the chances of midrange boost occurring?  I think it's a pretty safe bet that they do, as a rule, simply because of the size of them, and the treble problem I already mentioned. And if you are trying to compensate for too much midrange, then the rest follows. Most domestic replay systems these days seem to be midrange heavy to me as well - I haven't heard anything cheap recently that had anything like a flattish response - and they really don't suit the rooms they are in either.
    If you want to listen to material as it really should be, then you need to experience it live first, I'd say, and then do a direct comparison with what you can hear in your monitors. I'm fortunate - I can do this quite regularly with a variety of material. Do I tend to leave things as flat as they are recorded? Well, it depends on what it is. If it's in any way classical, then sometimes I look carefully at the bass balance, but generally I leave the rest alone. Everything else these days I just get to sound good - and that can mean all sorts of tweaks, depending on all sorts of things. More and more though, I've come to the conclusion that too much midrange isn't necessarily a good thing - but that's mainly because of the general lack of good reproduction equipment around these days.
    As for monitors and flatness - well that's not really an issue for most people, compared to getting their listening environment correct. If you have a pair of cheap monitors in a good room, the chances are that the results will sound better than an expensive pair in an uncorrected room - despite what all the monitor freaks on gearslutz might say. These days, even the cheaper ones can sound quite respectable. But flatness isn't an issue with monitors really - a decent impulse response, and low distortion are far more important. Chances are that if a manufacturer has got this right, the monitor is going to be suitably 'revealing' anyway - which is what a monitor is supposed to do.
    The one thing you do not do though, is EQ the feed to your monitors - that went out of fashion almost as soon as it came in - fortunately. You fix the room so that it's more truthful. If you EQ the monitor feed it will inevitably only sound good in one place in the room, and that's no use to man nor beast. The only decent things that proper room correction systems can do is equalise the immediate time response to take account of what's actually between the monitors and you - which if done properly can improve stereo imaging no end.
    So answers? Well not really. But at least I've explained a few (but not all) of the issues.

Maybe you are looking for