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bobkatz

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  1. Also, I play 7.1 sources here just ignoring the surround extra channels. I simply play the 5.1 as I believe that adding two surrounds together electrically into one channel will not authentically translate the 7.1 experience into 5.1. I just let sleeping dogs lie. So I'm not performing any matrixing at all in my Lynx mixer as I have a 5.1 system.
  2. The AES16 software can mix and matrix any combination of inputs to outputs. You can also load presets or scenes so if you want to change your mix you can do that by loading a new scene rather than painstakingly reassigning channels or levels. So, let's say you want to reduce a 5.1 mix to a 4.1. You take the center channel, reduce it 6 dB, and assign it to left and right outputs. Done! This is easily done in the Lynx AES16 mixer. Also, remember to engage the (24 bit) dither on any channels which are part of that matrix, to preserve resolution as you are then performing calculations. In this case you should turn the dither on in the center channel which has been reduced and is being sent to the left and right outputs. The existing left and right channels which are going to the existing left and right front speakers do not have to be dithered as long as they remain at 0 dB gain and assigned to their respective single outputs. There is, of course, a chance of overload when combining center channel information with existing left and right into the left and right, and customarily people reduce the digital levels of everything, say, 3 dB to protect from overload. There are meters and sticky peak indicators in the AES 16 mixer so you can see for yourself if you need to attenuate. Just pick the loudest, most percussive sources you have and do some tests on them. If you are playing material from movies mixes, usually the center channel information is dialogue so it may not add in together with the left and right front enough to overload. Very few music only mixes use significant amount of information in the center channel so in any case I estimate 3 dB will probably be enough reduction to "cover your ass". In that case, you will have to dither all the outputs as you will have to reduce all the outputs equally to retain the interchannel balance. The same goes for 7.1 you want to reduce to 4.1. You can matrix the two "leftish" surrounds into your left surround speaker and chances are they will never overload because rarely is there full level information in any surround channel. But again, play the loudest, most "percussive" material you have and inspect the peak levels in the AES-16 mixer to see. If you only have a couple of 7.1 sources and most of your sources are 5.1, you can leave the summing matrix engaged as the mixer is effectively noise-free and will not add any audible noise to the surrounds even though those channels are not in use. Lastly, be aware that if any of your 5.1 sources were derived from SACD, the SACD standard used an LFE channel that was not aligned to +10 dB. This is the most ridiculous, crazy thing in the world and defeated the whole purpose of the LFE channel as a special effects channel with more headroom, but anyway, what this means is if you are playing any FLACs which were digitally derived from an SACD, turn down the LFE channel in the Lynx mixer by 10 dB to hear the balance intended by the SACD producer. Provided he knew that fact and aligned his system correctly in the first place! A lot of producers making SACDs may have left their LFE channels aligned for standard 5.1 PCM mixes and forgot to realign them for SACD production. In that case as always, use your ears. If the bass part sounds weak, turn the LFE up. Watch those meters in the Lynx mixer, they will help you identify how and whether the LFE channel is being used at all. Different producers had different ideas of its purpose, and sometimes they used it to supplement the bass instrument unnecessarily, but only in order to see something wiggling in that meter so that they felt they were "accomplishing something". Something silly, but you as the consumer have to be aware of it to hear what the producer intended. Hope this helps, Bob
  3. Well, actually, working through the target curve is easy, it just takes time. I recommend you start with the philosophy that seems to agree with you. If you agree with my philosophy, that the target should be flat from DC to 1 kHz and then have a diagonal rolloff above that, then it's a pretty simple, but yes, time-consuming job. Since I started at 88.2 kHz for my measurements, and Acourate puts a dot in the target at Nyquist, then I use as my standard how many dB down at 44.1 kHz (the Nyquist freq) my target is set to. Then you can use a formula that will determine where to set the target for all other sample rates. FWIW, my target is -9.2 dB at 44.1 kHz, which ends up somewhere around -6 (if I recall correctly) at 20 kHz. You have to listen and change the target by 0.1 dB at a time until you are satisfied. Many people have been very happy using my target so if you like that idea, start with my target and you won't be more than a dB off, probably. As for managing decay. Absolutely. Your room should be as well managed as possible preferably BEFORE you start playing with room correction. If you know how to measure and interpret Schroeder curves and interpret waterfall measurements, do it, and fix it BEFORE you play with room correction. If not, hire an acoustical consultant, FIRST. Or learn about it, first.
  4. Acourate actually addresses room acoustics problems multidimensionally. It's not a complete solution, but it pays attention to far more than just spl response (frequency response). Acourate's psychoacoustic response curve avoids overcorrection, which is the bane of simplified spl-based solutions (which are performed by many of Acourate's competitors). As explained to me by Uli, his psychoacoustic measurement makes thousands of calculations, calculating the room's transient response. It only deals with the peaks which it deems to be psychoacoustically important. To my ears, it works well. The other thing which Acourate does is the variable FFT window, which is another form of psychoacoustic weighting. Some other systems (notably Audiolense) have a variable window, but Acourate's window appears to just work without user intervention. For more information on the advantages of a variable window, see the seminal papers by Jim Johnston et al, AES Preprints 7263, 8314 and 8379. As for the target curve, I can speak for the ergonomics of Acourate's target curve, it allows you to make the frequency response you desire and the attenuation you desire very easily. As for the reasons for a target curve, I'm not an expert on it so all I can say is that a user-defined target seems to be necessary. As the measurement system cannot take into account the following: 1) taste 2) room acoustics interaction with the polar response of the loudspeaker. E.G. the narrower the polar response of the loudspeaker, the less influence the side walls have in the overall frequency response. Thus we need to have a custom target for the particular room/loudspeaker/user combination. Hope this helps, Bob
  5. I have a Lynx AES16E card which installed (months ago actually) in Windows 8 with no problems. Perhaps you have the wrong driver set. I would follow the Lynx instructions literally. I think (it is possible) that for the first installation they may want you to install the driver before the card! I can't remember for sure but follow the Lynx instructions to the letter in cases like this. I'm also concerned that since you bought the server that perhaps there is some corrupted driver or other Windows problem. Did you do the Win 8 install yourself? BK
  6. To add to the confusion, it can be argued both ways whether some distortion in a system is more accurate than "no distortion"! The total harmonic distortion figure can be extremely low, but if it consists of predominantly the ugly harmnonics (7th, 9th, etc.) then it can sound harsh and less correct than a piece of gear with higher harmonic distortion but a more balanced mixture of harmonics. This is why you can add a piece of tube gear to a solid state chain and the sound can audibly improve. It could be argued that the tube gear is adding euphonic distortion, OR that the extra distortion of the tube is covering up some objectionable distortion from the solid state chain. That's one of many reasons why as of this moment there is no objective measurement method which cannot be questioned! We are still very much at a primitive state in terms of determining human masking thresholds when judging distortion, for example. Even though experts like Jim Johnston (co-inventor of the mp3) have shown us the way. So, just enjoy your sound systems, and realize that there are many truths, many of which are unknowable. If you make a change in gear, and it sounds better to you, it's probably better!
  7. That wasn't my recommendation, actually. I haven't used any Mac convolvers yet. Let me tell you a bit about the Sourceforge Convolver config format, which you can also read about at the Sourceforge Convolver website. 1) The measuring program has to create a filter file for each sample rate. These are usually in WAV format. 2) Some measuring programs also will attempt to create a file in .cfg format to conform with the Sourceforge spec. The .cfg files essentially tell the convolver which wav file to use for which sample rate. And there needs to be a separate .cfg file for each sample rate as well. cfg files can be edited with a text editor and it may take you a few hours to get one done. If your convolver conforms to the Sourceforge format, I have some example .cfg files that you can use to start with. 3) The convolver you use should include a log or a status file to show you that it is using the correct filter file and that it is switching properly with the sample rate. it's always good to check this file until you are comfortable your convolver is working properly. Never turn your back on computers :-). If not, yes, the upsampling option is a good choice and it may even be the better sounding choice, depending on which DAC you are using and how it performs at different sample rates. My newest Avocet DAC module internally upsamples to close to 8x 192 and has a very nice (chip-based ASRC) filter and is very well designed so I'm not sure I hear much of an advantage upsampling in front of the DAC or not. There was a time when upsampling in front of the DAC was definitely advantageous in my system. The differences you hear could be caused by: a) the quality of the filtering in the DAC's digital filter b) The quality of the filtering in the analog filter if there are different analog filters for each rate c) jitter or just d) Your imagination !!!!!! Hope this helps!
  8. Dear Dallas: Sorry to rile you. We got out of sync. I read your PM after writing my reply in this thread! You have supplied some important information about your NOS DAC. I'm still suspicious because anything that doesn't measure flat has to be colored. Many if not most loudspeakers measure a high frequency rolloff and so target design is often to mirror the slope of the loudspeaker's native response but get rid of the major up and down shifts. So I think even with a flat target and your rolled-off DAC you're going to have some issues unless your loudspeakers are flat on axis and possibly flat in the power response. Which ordinarily (without an NOS DAC) would be EXTREMELY bright! Anyway, let's please agree to agree that there are a number of controversial (third rail) subjects in the audiophile world, among them: 1) NOS DACs vs. Oversampling DACs 2) Linear phase versus minimum phase and Apodizing anti-image filters 3) and now, "To DRC or not to DRC". As a card-carrying audiophile up until about a year ago I was thoroughly in the "analog domain crossovers and analog domain EQ" camp. I built my own analog crossovers and even some filtering to deal with remnant low frequency artifacts. Out of necessity, because every DRC I had ever tried had some form of compromise or veiling. It was (and still is) the rule that DRC is not an audiophile-acceptable product. It's going to be very hard to convince most audiophiles that there is now even a system (Acourate, for example) that is the sonic exception that breaks the rule.
  9. Dallas, speaking of third rail, NOS dacs are the third rail of audioqphilia! Without getting into why I would never use a NOS DAC isn't it true that all current NOS Dacs are limited to 16 bit input? Well, Acourate Convolver (as well as the Convolver in JRiver) calculates at 64 bit floating point and dithers to 24 bit to feed the DAC. If you want to do drc, Dallas, properly you should use a DAC that will not truncate the high resolution information being sent to it. The issue with target roll off is the least of your worries, but yes, you can design any target. .
  10. That's one distinguishing feature of Acourate. There is no up or downsampling and Acourate works at the incoming sample rate. This requires automatic filter switching but the reward is increased sonic transparency.
  11. That's astounding! To be "pure DSD" from source to DAC you would either have to implement a volume control in the DSD domain or use a pure analog volume control post DAC. Miska, please tell us about your DSD drc solution. (So many abbreviations, so little time to learn them. :-)
  12. Yup: Acourate's 65K filter length has tremendous latency. More than a second at 44.1 kHz! This is not a problem for audio only listening. But as Mitch replied, if you convert Acourate's filters to .cfg format you can use them in JRiver's Convolution engine, and latency is completely taken care of. It's a wonderful pleasure to watch and listen BluRays with 5.1 surround fully corrected by Acourate. All sample rates from 44 through 192 (and possibly beyond but I've never tested) are transparently taken care of.
  13. I'm using Acourate at rates up to 192 right now, no problems. As for DSD, there is no hope for Acourate correction to work natively with DSD. All of these correction systems work in PCM and cannot play native DSD without converting to PCM first. So I lost the ability to play my SACDs when I installed Acourate. When I get the budget I'm going to try out the Oppo BDP-93 or possibly the 103 player with the Audiopraise Vanity converter. Since it effectively can "upsample" 64x DSD to a high rate PCM (176.4 kHz) with very low distortion, it is possible it can do that transparently or reasonably transparently. I hope to get that going in Surround to play my Surround SACDs in 176.4 kHz PCM with the Vanity mod. There will always be a sonic difference, but I've never considered 64x DSD to be the be-all-end-all. It's a bit "softer" sounding than a high rate PCM original. There's nothing magical about SACD, but it's a pleasant format and certainly better than 1644 CD!
  14. Regarding tweaking, what I found is that Acourate is the first DRC system I've tried where I was not tempted to further tweak the settings after they were made. (except for the Target, of course). Almost every other system I've tried overcorrects in some frequency range or another. Acourate does not overcorrect (or apparently, undercorrect, either). I've not had a suspicion of a bass note out of place since I got Acourate going. I am running extensive trapping in a well-designed room with a reflection-free zone, so your mileage may vary. The worse the room, the more likely you're going to find some resonant notes or missing notes, as you mustn't rely on DRC as your correction. DRC should be the icing on the cake in a well-treated room.
  15. It should be possible to supply filter files to any player that has a built-in convolver. Acourate generates standardized filter files and with some work you can also create .cfg files in SourceForge's convolver format.
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