Klewless

I can vary the delay (digital) to my sub via my Rotel Home Theater receiver. I can only reverse the phase on my Rel subwoofer. I understand that some subs can continuously vary the phase. My confusion concerns whether or not amplifier DELAY is the same thing as subwoofer PHASE. I want the sub to wait for the Khorn bass. Thank you for helping me understand the difference. John

Yes, thank you. I have 350 or so albums to capture. I currently have them stored alphabetically in six major groups. Been that way for over 50 years. I want them to be computer sorted likewise with whatever constraints foobar might need to make it easier to relate the computer location to the physical location. Does this make sense to anybody? I can't seem to find out just what these media players (I have looked at a lot of them) need to function "properly". I most likely will still play the entire record but the ability to skip around is something not easily done with a turntable, so there is somewhat of a learning curve for me. I am hoping to avoid "filing mistakes" espcially if others have worked through all this already. Thanks again for responding. Any operating tips anyone can share would be greatly appreciated. John

What kind of folder structure do you use for foobar20000, eg. album, artist? ... etc I am starting the long process of capturing my 350 albums and hope to do it "properly". Lots of confusing issues to me anyway since there are a lot of many-to-many relationships involved. I find it very difficult to get questions answered on the net or to get a help file that actually helps. I did stumble onto how to get foobar to display my album art. And I just discovered how to preserve the order of tracks (the same as on the phono record) for foobar. Any kind of operating tips would be greatly appreciated. John

Just how much music is taken away by this format? In your opinions, of course, is it "good enough" compared to the cd format? Thanks for any constructive comments.

Hi Dave, I got my M-Audio USB Transit to work! Don't understand how but I got there. The fine print says it will not go higher than 48K on the analog input! However when I disabled output in it's control panel, it allowed me to choose 88.2 K for the input. Unfortunately Magix would not read beyond 48K, so I was not able to test. It did work for 48K though. Were you talking about Sound Forge Audio Studio 10? If it is around $80 I will get it. Thanks for your input. Now I will download the free trial and see what happens. The next big question concerns Media Players. Are they accurate in data handling? I would like to use FLAC for storage and playback. And if really want to have my cake and eat it too, I can just play my records as they are! But I do like the ability to sift through my collection in ways I cannot do now. Thank you again. John

Hi Dave, Thanks for responding. I was trying to decide if my two pairs are essentially equivalent in performance. If I should do a CD, is the conversion from 24-to-16 accurate? That is why I was considering 16/88.2. Right now I am using Magix Cleaning Lab and it does not seem to support 88.2 So I should probably consider another ( USB) converter. Actually I have an M-Audio Transit but cannot get it to run under Windows 7 (32 or 64) inspite of getting the latest updates. If someone can help me get it to operate I would prefer to use the M-Audio. I have seen many 16/44.1 references to wit "that is all one needs". I could be very happy with 24/88.2 as you suggest but will need something to replace my Magix Cleaning Lab to capture my 350 albums. Thanks again for helping me "see the light". John PS. How does one use DSD?

I am aware of the endless discussion of this topic. Therefore ... My question is "which is better considering the affordable hardware implementations (CD/DVD and computer sound cards)". I may or may not make CDs after I capture my albums. Stated differently, I am not interested in theory , only which gives better (if any) final results. In short, which of the two choices here is more accurate 1. more samples or 2. finer resolution. Again not in theory but in real world devices. Thank you for your opinions. John

I use a Rotel Home Theater receiver's decoder (Neo and Dolby Digital) to derive a center channel. The one feature I really like is that the CENTER information is REMOVED from the Left and Right speakers. The receiver first converts the incoming ANALOG signal to DIGITAL! My question about center channel is whether or not the signal I get now is the same (or different from) the derived SUM signal from the Klipsch resistor network. Thanks for any clarification you guys can provide.

"So why did PWK use the long constricted throat on the K400? Why does any design call for it? Does it provide more gain? If it does, I think it does so at the expense of coloring of the sound. " I believe the longer mid horn was based on lowering the cutoff frequency from 500 Hz to 400 Hz. Standard "horn theory": lower freqs means longer wavelength means longer horn. Tradeoff means more beaming of the high frequencies.

Can any of you post the crossover schematic and acceptable drivers to build the Academy? I want to build myself three of them for my system. Thanks for any info that is legal to share.

Sorry about my ignorance, but what is an anti-cable?

Anyone ever consider a Constant Directivity Tractrix?

Google "media player" or "music server" (you don't need to put the quotes in Google) and follow all kinds of links to learn a lot. Lots of stuff to learn. Do not be intimidated by the complex issues you will find online. Consider it an adventure. I am currently doing the same thing.

I got to put my 2 cents in on this one. My opinion (of course) is that as far as your ears and source material are concerned there is no point in going beyond 20K. However for electronics (amps) there may be a valid reason for going 100K or better because of the feedback from output back to input. It seems reasonable to me.

Thanks for reply. I stumbled onto it in some blog about doing HT media. He also loved Sony Sound Forge. I downloaded their "free trial" and they insisted on registering it online, so I just forgot about trying it out. In my mind, free has no strings attached. One of the things I am struggling with is format I should copy my records into. I want to avoid as much sampling and re-sampling that apparently goes on. I believe that means use ASIO and have the right format for the sound card.

Anyone familiar with the Juli@ from ESI? I am building an HTPC to capture all my phono records and am considering this sound card. The software I am considering is Sony Sound Forge Audio Studio. I would appreciate ANY opinions or alternatives. Thanks Web esi-audio.com/products/julia/

It is 16 bit 48K WAV. I do not know what iTunes does (or does not) do to it. For all I know, it might even down convert it to 44.1 K.

Has anyone posted the acoustic impedance plot of the Khorn and LaScala bass section?

Thank you for responding. I have tried the optical output to my receiver but got lots of loud snaps and pops (analog worked OK). Obviously something is wrong but I have not been able to discover just what was happening. Thought about buffer over/under runs or overloading input. Still lost on just how computer files are actually output.

I am using an M-Audio USB soundcard to my receiver. It has both analog and digital outputs. I can control the volume at the computer and the receiver on BOTH analog and digital. The digital part is what confuses me! Shouldn't the DIGITAL output level be constant? I am using iTunes for the source. All I want from the computer is to stream the digital signal (WAV) from the harddrive untouched to the receiver. Is there any other device other than soundcard? Thanks

Does your manual tell how / when down sampling occurs? I believe that is an issue only on the player's digital output. With the analog (usually 2-channel) outputs sample rates are not involved.

I do appreciate the many variables which go into the design of ported speaker enclosures. I am looking for a simple answer to the act of increasing the enclosure volume and retuning the system to the original frequency. Would that INCREASE or DECREASE the bass response at resonance? I am guessing minor volume change would have minimum impact (maybe not even noticeable) and something like 50% increase would produce less bass due to lower driver loading. Thanks for your response.

I use a sub with my Khorns set to the lowest frequency possible (40Hz). I also set the Khorns as small. My reasoning is to keep the sub from influencing the Khorn sound as much as possible. The other reason for limiting the bass is to eliminate woofer motion below the Khorn cutoff frequency which tends to clean up the sound at higher woofer frequencies.

I vote for 1--LaScala with sub, 2--cornwall without sub, 3--heresey with sub. My experience has been that proper placement is more important than the speaker within reason. Shifting them slightly forward and/or sideways usually work well for the image. I am currently doing what I preach with Khorns with sub in a very bad room (long and narrow like a trailer) and am very happy. I aim the horns to cross about a foot in front of my face (therfore the Khorns touch only the outside walls).

Thank you all for your input. I expect to capture at 16 Bits and maybe at 48K so I can still brag about being an "audiophile". Some of my albums have scratches which I plan to learn how to clean up ( have Magix Cleaning Lab software). The playback will be iTunes or Win Media Player, unless someone can suggest a better player. Right now I am experencing a playback problem. I want to send the digital signal from my PC to my HT receiver. Unfortunately the only digital output I currently have is OPTICAL on my USB soundcard. Sometimes during playback I get a loud noise as if something got lost or ???? The one issue which puzzles me is that iTunes can control the volume to the sound card. I originally thought I was overloading the input to the soundcard so I turned the volume from the PC way down but the noise still shows up randomly. I do not understand how the volume can be controlled at the OPTICAL input (which I expected to be at a constant level). Maybe it is a buffer size issue?? The music path is USB harddrive to the PC and USB to the soundcard.