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sfogg

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Everything posted by sfogg

  1. "OK, so where is all the material? " The whole point is you don't need all new material for music in surround. That is the point of things like DPLII, L7 and Trifield. It works with existing music. As for other material seems to me there is more and more lossless multi-channel coming out every week then there has ever been available before. "I can make surround recordings that will transport your mind." Perhaps, but I have also been listening in surround for a long time now. "Unfortunately, I can't get them to you." Why can't you get them to me? It isn't terribly hard to do. Send me five wav files and I'll send you back a Dolby Digital encoding of them on a DVD that would play in any system. Or you could use one of the open source codecs to do the same thing. Check out flac for example which would have higher quality then Dolby Digital. The tools are available, use them. "I wouldn't do it that way. Adding metadata to digital data without altering the original data isn't rocket science. " I'm talking about when the distributed audio doesn't match the playback system. The audio is going to have to be changed to accommodate this in any reasonable scheme. Unless you think the studios are going to make a couple of dozen mixes of the same album for all the different possible system configurations that are out there. If so how exactly are you planning on delivering this sort of thing which is going to take up an enormous amount of space and bandwidth? "No idea what that means or the context. Of course it is. " If you don't understand the difference between lossless compression(Winzip) and lossy compression(AC-3) you really shouldn't be commenting on this. "If you want to believe our current standards are the end all, fine." I don't think they are the end all. I do however think they are far better then you give them credit for. To me it looks like you are hung up on bad mixes and blaming that on the delivery format. It isn't the delivery format that you really have any issue with, simply the engineering choices of those making the music. Instead of trying to come up with a delivery format that is never going to go anywhere you have a far better chance of showing what can be done with existing delivery formats and pushing for more quality music. "Feel free to tout the present system and perhaps you think those of us who find it lacking in ability to recreat an acoustic space/time event are just too picky." You haven't heard 2 of the 3 systems I'm talking about. Your chosen speaker setup is not taking full advantage of any of the three I have mentioned and IMO has other problems. You aren't really in a strong position to comment on a supposed inability of them to recreate an acoustic space. When I play back music the acoustic space changes from small intimate settings to large halls and everywhere in between. If you aren't getting that perhaps you need to try something else. For S&Gls I can even alter the acoustic space of anything that occurs within my room. Want it to sound like a large hall, I can do that. Shawn
  2. " If that's what my tubes are doing, I'll take it any day over a stale sounding AVR. " Not your tubes, just the L/R speaker trying to phantom image a front and center vocal. Two speakers playing identical material interfer with each other and cause many small cancelations between them. There are also four differing arrival times for vocals when there should only be two. All this is unnatural compared to the rear world, voices are always created from a single point in space. That vocals have always sounded this artifical way in two channel can sometimes take a little unlearing when moving beyond two channel. Compared against the real world though it is obvious which is more realistic. There is plenty of surround equipment well beyond a stale sounding AVR. Get a pre-pro and use it with tube amps if desired. I ran tubes up front in my systems for many years. Shawn
  3. Dave, "What I am doing is pointing out the issues of the current system which the majority agree is not working. " Not sure who the 'majority' are but the existing surround system systems have sold far more audio equipment in the last 10 or 15 years then anything stereo only. "The metadata would be added "on the fly" at the source for the kind of recording I do, or in the mix for created music or movies. But it would not alter the digital stream of the music itself. Every consumer would get a bit for bit master and that is what they would hear. The "processing" would be simple routing information like a zip code. No change to the information but insurance that it gets to the right address." Sorry, but that system would still need to alter the data. Say you build a mix that had chunks with 7 channels of audio in it. You play it back on a 2 channel system. Either you are going to perform some type of downmix to two channel (which involves altering the data) or you are going to be playing it back missing a bunch of the audio. Likewise if it only had 5 channels in it but was played back on a 7 channel system there needs to be a method of utilizing the playback system. "one still needs universal physical media and playback systems to deliver high definition surround audio experiences to users in a low cost, easy to use medium. " How many people do you know that don't own a DVD player? "Further, you also then had to deal with AC-3 rights and Meridian Lossless Processing, a system no more complex than Winzip but enshrined as a "gatekeeper" cost to protect music interests." Dolby Media Encoder SE will deal with this for under $600. Lossy AC-3 is *dramatically* more complex then Winzip BTW. It has to analyze the data using a perceptual audio model to throw away data it thinks is not needed as well as to look for correlated information between channels that it can reduce for smaller sizes. Lossless Meridian Lossless Packing which is Dolby TrueHD is closer to Winzip since it is bit-perfect. It is already built to scale to handle increased number of channels using the equivalent of metadata. The idea of using extensions within data to enhance audio formats at a later data has also already been done. Look at how DTS added ES and 96/24 support to their core audio. On playback it still means needing upgrades to the receiving equipment to understand how to utilize the extensions. From talking to DSP engineers that dealt with this this method also requires MORE DSP power then just decoding a format that didn't use extension data like this. Shawn
  4. " The quotes are to remind me to point out I am not suggesting the processing of the audio streams, but simply reading metadata and routing them accordingly" How can you take 4 channels of input and 'route' the audio over x number of speakers without using a bunch of processing? Routing is just another word for steering...all the metadata does is remove some (certainly not all) of the assumptions about where you might steer the various pieces of the audio with some clues on where to steer it. The steering still needs to occur which involves all the same processing as something like PLII or L7. That sort of system really exists already but it doesn't need the metadata. DPLII and Logic 7 are also both encoding formats as well as decode/playback. You can take a multichannel input and encode that down to two channels of audio. DPLII and L7 decoders/processors look for the encoded queues on where to steer the audio back into multi-channel. I've taken multi-channel audio and L7 encoded down to two channel and used L7 to get back to multi-channel. It sounds very very close to the original discrete multi-channel mix. And just like you propose if you don't have any processing it is a perfectly normal sounding 2 channel audio. If you have 4 speakers the L7 decoder handles that, if you have it handles that, if you have 7 it handles that too. Shawn
  5. "....correct imaging/soundstage." Play a live recording over two channel. It does not have correct imaging, the audience isn't on stage with the performers. The halls ambiance is being reproduced from the completely wrong direction. In the real world you can listen to a performance and move your head or whole body without having the imaging shift side to side. In the real world the instrument we are all most familiar with (voice) isn't riddled with comb filtering artifacts. Music in surround (wether recording that way or processed into surround) can improve on all of the above. For example try Dave Mathews 'Live at Luthor College' in 2 channel. Then try it in a good surround setup with something like Logic 7 Music. The difference is *stunning* (in every way) and very very much in favor of being reproduced in surround. Shawn
  6. "Been a while since I paid attention to Ambisonics, but I do not believe it to be a completely digital situation." From what you are describing it sounds very much like what Ambisonics aimed to do. http://www.ambisonic.net/pdf/ambidvd2001.pdf Shawn
  7. " I've read of some work being done to encode tracks with digital data to route them to a speaker placed as close to where that sound should originate as possible. With such a system, one might have 15, 20, or more speakers with each one further defining an accurate sound field.This MAY happen, as the proposed technology doesn't require any specific number of speakers or amps so doesn't really mandate anyone do anything. " That sounds like Michael Gerzon's Ambisonics. Meridian processors have that but there aren't a huge number of recodings that are recorded in Ambisonics. The ones that do can sound very nice but there is a flaw in Ambisonics. It is trying to reproduce all the sounds for a single point in space. We hear from two points in space. Gerzon is also who did Meridian's Trifield. On nice thing that came out of all that work was the Ambisonics Soundfield mic can make really good recordings without the Ambisonic encoding. Cowboy Junkies 'Trinity Sessions' is likely the best known example of this. Sounds great in 2 channel sounds downright spooky in an excellent multi-channel system. Shawn
  8. "When it's music only I do that in 2 channel - it's all tube so it's hard to step away from that in music only applications" You can do tube processing and multi-channel together too. Just need to find one of these: This was from Jim Fosgate. Jim Fosgate is the guy who built the circuit (with tubes in the analog domain) which become Dolby Pro Logic II. Fosgate built a limited run of his analog tube version as shown above. Shawn
  9. "However, the fact is I've not heard one yet that beats the Hafler for purity and putting stuff where it actually should be without sounding processed, and I was not referring so much to the systems as to the mixing" That sounds like it is more the limits of Hafler masking the mixing choices that you don't like compared to a problem with the modern system which are more revealing of mixing choices. IME Hafler can be fairly pleasing (it was what I started with first too) but the lack of hardly any channel seperation in the rear channel can be a problem in most setups due to the Haas effect. Ambiance being totally correlated (mono) is unnatural compared to a hall. You should listen to Trifield sometime. It works on a different manor from DPLII and Logic 7 and is an unsteered system at least on the front three channels. Up front it sounds very natural but the lack of steering limits the envelopment in the surrounds and the surrounds are too correlated which again sounds unnatural compared to a hall. DPLII and L7 both can be adjusted quite a bit to taste too. In DPLII set Center Width to Max and it won't steer from L/R to center. Panorama off and Dimension to front. Surprised you like Neo. That one is rife with real steering errors that have nothing to do with the mix itself. Shawn
  10. "I've heard only a few examples of modern "steered" and logic controlled multi-channel that I consider as good. Not sure why engineers haven't quite gotten it yet." IMO they have. Check out Logic 7 Music (the Lexicon version not the dumbed down H/K version), Trifield (Meridian not Yamaha) and Dolby Pro Logic II Music for some very good examples. Shawn
  11. "Is it because Sunfire rates these amplifiers at 1 kHz rather than 20 Hz - 20 kHz" That is probably the difference. It is easier to play just a 1kHz tone then it is full bandwidth. Also consider the difference between 343w and 400w of power is almost meaningless, it gives you less then a dB greater output. Shawn
  12. Why would you want to? If you are going to use it as a tweeter you could get a dedicated tweeter horn which will be physically much smaller and likely have better dispersion up high. Shawn
  13. It is a persepective trick. My take on it. The trough goes downhill. Look at the height of the sidewall on the trough. It gets skinner as the water flows. This along with the perspective of supports give the illusion of height.It isn't tall, it is relatively horizontal moving back away from the camera. I think the base likely has a pump which pumps the water up to the outlet which is arranged such that from that specific camera angle it looks like it is flowing from the 'top' of the ramp. if you could see this from a different angle it would look totally different. Its like those 3d chalk drawings. From dead head on they look amazing, from the side they look extremely elongated. Shawn
  14. Al, "aren't we all proud of the efficiency of out horn speakers? It seems a crime to cut it back." It is a trade off, like everything else. If you want to do it passive you loose some system efficency but in its place you make big gains in the power response of the system. Some are willing to make that trade off, others aren't. If you want both you go active. Shawn
  15. That has been my goal, not the 'they are here' I am going for 'I am there.' The problem is this is extremely difficult to achieve in a 2 channel system. The halls sound is originating from the completely wrong direction. Shawn
  16. "That limits how much you can cut down the extra efficiency (above that of the woofer) of the high section." It only limits it if you aren't willing to lower the overall system efficiency. Like you said if you have a woofer that is 104 dB/w/m and a high frequency driver that is 111 dB/w/m you have 7 dB difference you can use as EQ. If you are willing to drop overall efficency to say 98dB (by padding down the woofer) you have 13dB difference to use in EQ. Shawn
  17. Dave, "Using a single driver for the top end is basically the same premise as as a time aligned top end. In a time aligned top both drivers are same distance from the listener or a delay has been used so the sounds from the individual drivers reach the listener simultaneously." Not quite the same. When you electrically or physically time align two drivers you are still aligning them for only one plane in space. Anywhere not on that plane is still not time aligned and will therefor comb. What you hear at the listening position is a composite of what occurs on-axis as well as off-axis. It is why some designers work very hard on the power response of the system. If you are combing off axis that still will alter what you hear at the listening position. When you replace two drivers with a single driver you are eliminating that combing everywhere in the room. Shawn
  18. "Not possible from what I am aware of regarding physics. I can understand the idea of being more capable of louder LFE with more transducers, but going deeper with more...please explain" They couple better at lower frequencies (just like multiple subwoofers will do) and the additional output ability (roughly 15-20dB going from 1 to 12) means you can tailer their response more with a little EQ. There are also physical mods you can do which will lower their resonant point, though I never bothered with that. "I watched batman dark knight at reference levels and never noticed any issue with the single aura pro in my couch running out of gas. " Good, enjoy it. "Maybe my couches are more lightweight than most" Or your floor is different then my floor. A floor that gives needs less then a solid floor. I've been using these for about 12 years, one wouldn't get it done for me. Shawn
  19. " I honestly have no idea how anyone could need more than one of these per couch," More let them go deeper and to not overload on the deep/loud parts. Like you said it also depends upon what you are mounting to and what that is sitting on. That couch was on a cement floor. The more springy the seating the easier it is to get the effect which is why they now make isolators for seating. If you have the gain turned up to high they can be very annoying. The idea is to have them compliment the bass, not act as a distraction. Shawn
  20. The various shakers can be a lot of fun. Several systems ago I had 12 of them mounted in my couch. Totally changes the T-Rex attack scene in Jurassic Park. In the theater I have 2 of them per seat now. A really fun scene to demo the shakers to your unsuspecting victims (guests?) is on the bonus disc of The Incredibles. Try out "Jack Jack Attack" at THX reference volume. This scene starts out very quiet but have had a number of people literally jump out of their seats when it all kicks in. Just be sure to treat them like a subwoofer with low passing them and if you can add delay just to the shakers they integrate with your main bass better. I used a programmable trigger to have them only turned on for movies. Shawn
  21. "It looks to me that it is. At least one is required to make the other. You are moving a small window in time (a segment or interval of time) and computing the spectrum during that interval. Each segment is converted to a spectrum and displayed one in front of the other which looks like the waterfall. " OK, from that perspective a waterfall could be considered sort of like a composite of dozens or even hundreds of windowed responses arranged in chronological order. But I think( could be wrong) that the mechanics behind the measurement are a bit more then that. If you measure frequency with a very small time window (one of those composites) you loose low frequency resolution, the waterfall doesn't have that issue. My main point was just windowing one measurement and looking at it is a 2d display, a waterfall in one graph shows dramatically more information about the behavior of the driver or system or room in the time domain. Shawn
  22. Al, You can get a simple USB soundcard to get audio into your computer. "It has and adjustable window for the time capture feature but I don't think it will look at a narrow enough time interva" A waterfall isn't the same as windowing a measurement. Windowing a measurement is basically averaging over the window of time you select. That isn't what a waterfall is about. A waterfall is showing your the response of the system in 3d ( over time), not averaging time and displaying in 2d. As an example of what you can learn from waterfalls check out this page: http://homepage.mac.com/marc.heijligers/audio/milestones/acoustical/dampening/dampening.html Waterfalls can show you problems that are harder to see in 2d. For example I think someone talked about removing ferrofluid in their driver and seeing a little bit better frequency response up high. That is certainly possible but the ferrorfluid also may have been damping the vc/diaphragm and removing it may have added resonance issues. The waterfall would show if that is the case or not. "Shawn, I think the SD380 analyzer had this capability as an option. Does the you have / had have it?" I'll have to check, if it does I have never used it on that. I always did waterfalls with ETF. Shawn
  23. Al, Grab http://www.hometheatershack.com/roomeq/ And you can do waterfalls on your computer. Shawn
  24. Al, "When Lee does the polar plot I can display the waterfall data from that data." I'm not sure what you mean here? The waterfall Mike was talking about has nothing to do with polar response. It is more like a 3d frequency response... showing the amplitude/FR of the system over time. This is very handy for seeing resonances in the system which might not show up in a 2d FR graph. The reason understanding resonances can be important is the system is literally playing a note when it shouldn't. That ringing maskes detail. Shawn
  25. Mike, I don't think I posted all of that, not sure I can find all the old measurements of the system. I ended up crossing at 500hz, from talking with Roy between 500-600hz the LaScala bass bin has about 90 degrees coverage so that is a good area to match the coverage to the K510.The K510 starts unloading below 500hz so it wouldn't have gone lower without EQ. I did knock down the big peak in the LaScala (about 10dB around 170hz as I recall) but didn't completely flatten that out as I preferred the sound with a couple of dB lift in there. I cross to the subs at 70hz. For the K510/44xt what you see in those graphs is how I left it and have been very happy with it. Might not work keeping it that flat in all rooms though, my room is extensively treated and very dead. I can tweak it with the bass,treble and tilt control on my Lex. if desired but almost never touch those controls as I just don't see the need. Shawn
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