Hey Guys and Gals:

Something you might be interested in...

A couple of years ago I got into a (heated) debate with a friend regarding the then new stereo 24 bit/48K audio DVDs coming onto the market. He swore that the sound of the 24 bit "wipes out" the cound of the same program on 16 bit CD. He swore that the difference was "enoumous" and that he could tell instantly which was playing.

I couldn't leave that hanging.

Having a lot of time on my hands, I took the title track from the audiophile re-release of "Kind of Blue" by Miles Davis and imported it into my digital audio workstation (a professional ProTools 24 bit system). Using professional mastering software, I made two additional versions of the track; one bounced to disc as 12 bits (the lower 4 bits set to 'zero') and one bounced to disc as 8 bits (the lower 8 bits set to 'zero'). Simultaneously, I applied noise shaped dither to both versions at their respective bit depths: there is _no_ information below the 12 or 8 bit level on those versions. (Noise shaped dither is less audible than straight dither, and linearizes the least significant bits better). I then took the original 16 bit version and the 12 and 8 bit versions and laid them on my workstation's screen, and sliced them into sections. I took the different bit depth sections at random and edited them into a continuous file at 16 bits so that it could be burned to CD, noting the time where the switching from 16 to 12 to 8 bits occured relative to the display on a CD player, then burned the result back onto CD.

Hey, if 24 bits 'wipes out' 16 bits, then hearing the difference between 16 bits and 12bits or _8 bits_ (!) ought to be like shooting fish in a barrel, right?

So I gave him the CD and he said.........(-:

Anyway, if anyone wants to have some fun, I'll make five of copies of this CD and pop them off in the mail to anyone interested, with the provision that they in turn send it along to anyone else interested, after listening and posting your impressions here. You can also feel free to rip a copy of the CD before passing it along.

Of course, I am not going to reveal right now the timings of the changes....but I will, in due time (-:

Any takers?

PS: You can give me your address by Email so you don't have to post it here.


[This message has been edited by soundhound (edited October 10, 2002).]

Sure write me at rbessinger@deloitte.com and I will give you my address. I think this is an interesting experiment.

I'm in! Email Address: Dollar2@attglobal.net. or maybe you should post your address here and we can send you our mailing addresses. I look forward to it.

charlie@bgtree.com

That does sound pretty intriguing -- I'm at gonk@prillaman.net . If it's a single track (and not too big), I might even be able to scrounge up space to host the WAV file.

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gonk -- Saloon Links | Pre/Pro Comparison Chart | 950 Review

I'll give it a shot.

brianca@ud.com

brianca..

OK, Brianca makes five people, and I have 5 CDs. I'd love to make more, but I've got to feed my cats too!! I've instructed these five people to forward the CD on to anybody else, via this forum after they listen or make a copy. Sorry to have the rest of you wait, but us Outlaws are used to waiting, aren't we (-:

Maybe Gonk can host the file??


[This message has been edited by soundhound (edited October 09, 2002).]

- practice makes perfect?

I'll see what I can do about hosting the file, at least for a couple weeks.

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gonk -- Saloon Links | Pre/Pro Comparison Chart | 950 Review

By the way, for those of you getting the CD - I'll give you a hint: The song, "So What" starts out in 16 bits, unaltered, and then goes to ?????

Also, I might have been unclear, using the word 'truncated' for making the 12 and 8 bit files. Actually, the software takes the least significant bits of the original file, and 'folds them into' the least significant bits of the new file, during a digital bounce to disc, while adding the noise shaped dither.

[This message has been edited by soundhound (edited October 10, 2002).]

I can't stand the suspense... was your friend able to tell the difference? Perhaps not accurately, but at least that a change occurred?

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Matthew J. Hill
matt@idsi.net

well, yes and no...........in no particular order.........

gonk: are you hosting the file?

Otherwise, where can I get it?

BTW, this CD on SACD is one of my favorite demo SACD discs....

I can host it as well. As soon as I get the CD, I'll pull an image and post it for those that want to grab it and burn their own. I'll make myself a copy and send the CD on to the next person that wants it.


brianca

Great! The song is about 9 minutes long or thereabouts, so the file will be pushing 95MB - but I hope you can do it.

I just addressed the CD for you, and will be taking it to the Post Office tomorrow morning.

I've started clearing some room for the file, and will host it when I get my copy of the CD. I'll stick a link to it here when I get the file up.

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gonk -- Saloon Links | Pre/Pro Comparison Chart | 950 Review

Great! I mailed the CD this afternoon.

Although this is an interesting exercise, it really isn't the same as comparing digital audio at different bit depths. 24 bit records don't just add 8 more bits below the current 16.

By truncating the least significant bits, your are cutting information out, but only certain parts -- the quiter parts. A true 12 bit recording doesn't totally lose some of the sound, it just has effectively less resolution across the entire sample. So, by taking a 16 bit recording and truncating the bottom 4 bits, you are keeping ALL of the resolution contained in the most significant bits. Take a single piano note, struck loudly. By truncating bits, the beginning will likely sound identical, but the 'fade' will stop suddenly once it reaches a certain level. If the entire note was done at 12 bits from scratch, you wouldn't get that sudden crop, but rather less resolution in the dropping sound levels.

Think of a still image, with 24 bits per pixel (8 bits each for RGB), then truncate 6 bits off to make an 18 bit per pixel image. If the original image consisted of only very bright pixels (narrow dynamic range), you probably couldn't tell the difference. If the image instead had a very broad dynamic range (bright sky fading to a dark horizon), you would lose fainter details. If the image were resampled to an 18 bit depth, instead of simply cropping, some of those lost details would be retained, at the expense of banding as the sky faded.

Stereophile magazine had a good CD out a few years ago for TRUE comparisons of recordings at different bit depths, called Testmasters. They recorded a solo piano along multiple paths at the same time. Same performance, same microphones, same levels. Along some paths, analog mastering was used. Various digital paths were used, including straight 16 bit, 24 bit with dithering, various dithering techniques, etc, HDCD, and so on. It's easy to compare each technique by skipping tracks around.

It is also easy to tell the difference between straight PCM and 24 bit dithered down to 16 bits -- not to mention the differences that would most likely be evident if the experiment were re-created today with a medium like DVD-A, which would allow true 24 bit recording to be directly compared.

Bigmac:

Read my 3rd post above. I didn't just throw away the lower bits - it was done the proper way with professional mastering software and a ProTools digital audio workstation, by taking the lower bits of the original file and folding them into the lower bits of the new file during a direct digital bounce to hard disc. Noise shaped dither was added at the same time at the appropriate bit depth. Trust me, I work with digital audio daily (yes, including 24 bits, multichannel) as a Music Editor for motion pictures, and I know what I'm doing. This IS a valid test. I don't have any particular agenda here, other than to give people who wouldn't have an opportunity otherwise to hear what these lower bit resolutions sound like, and maybe they'll have a little fun in the process. Why don't you keep an open mind and download the file when it becomes available, or get it from one of the people I have sent it to when they are finished with it.

I didnt mention on my original post that this track is just part of an entire CD I made, containing a variety of material, and YES, I made ORIGINAL 16, 12 and 8 bit recordings of a piano for the CD. And yes, I made 24 bit ones also, as my workstation is capable of 24 bit resolution. I simply made the Miles Davis available because it is widely recognized as an audiophile favorite demo piece that many people already posess, so they can compare.

[This message has been edited by soundhound (edited October 10, 2002).]

I see bigmac's point that if the 4 (or 8) lsbs were simply tossed away (truncated), that would not the same as orignially recording at 12 (or 8) bits.

To illustrate this, say you have two tones. One tone is large in amplitude, consuning a large percentage of the recording medium's dynamic range. The other tone is small, by itself consuming less than the range of 4 (or 8) bits, the number of bits you are later going to truncate. If you record the big tone only, then both tones together, and then the small tone alone, then simply throw away (THIS is truncation) the least significant 4 (or 8 bits), the playback result will have the large tone in the part where just the large tone was recorded (with distortion due to the truncation), the region recorded with both tones will have both tones (with distortion due to the truncation) but the region recorded with the small tone alone (which had a recorded amplitude less than the number of bits that was truncated) now has nothing.

In soundhound's orignal post in this thread, he said he truncated the least 4 or 8 bits. From what he said later, it sounds as though he did not actually truncate (chop off). Beings I am not a recording engineer, I do now know what procedure or process would be used to go from 16 bits to 12 or 8 in a non-truncating way. Of course converting back to analog and then re-digitizing with a 12 or 8 bits word length would do the trick. I have no doubt that there is a digital process that can take one from 16 bit word space to 12 or 8 bit word space(without truncation) without going through the intermediate analog state.

I remember years ago when I purchased my first computer sound card (a Turtle Beach Monterey, with I stil have but don't use because it is ISA and non-plug and play). I played aroung with recording, via the line input, various snippets of sound from an audio CD player. I recorded in 16 bits and 8 bits and compared the two. I was shocked at how good (compared to my expectations) the 8 bit recording sounded. I was expecting the 8 bit recording to sound much worse that it did.

So, when soundhound says that he has propely went from 16 bits to 12 or 8 (non-truncating), I suspect that it will be not as easy as we might think to tell (by listening only) which section of music was recorded with how many bits.

Paul

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the 1derful1

[This message has been edited by Paul J. Stiles (edited October 10, 2002).]

Geez, I'm sorry I used the word 'truncate' in my first post!! I have since corrected it. (-: I explained this further 3 posts later. You are quite correct that if I _did_ simply chop off the lower bits, I would be throwing away valid information in the original file. The way I did do it in essence 're-recorded' the original file digitally (but with _no_ A/D or D/A conversions taking place) at the new bit depth as a new file, while adding dither.

I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.
I will _not_ say "truncate" again.

(-:


[This message has been edited by soundhound (edited October 10, 2002).]

Ok, I got hung up on the original 'truncated' description, taking it as if you were just dropping the least significant bits.

This makes more sense as a more valid comparison -- but is still not quite the same as comparing a 16 bit track versus a true 24 bit track.

Quite right, but I wouldn't be able to use a CD as a distribution vehicle otherwise, and a lot of people wouldn't be able to hear it. The test is still valid as a comparason between the bit depths represented. I do have many 24 bit, multi-channel master copies of music scores for a lot of films I've worked on, and ones you probably have on DVD.

[This message has been edited by soundhound (edited October 10, 2002).]

Sure yah could -- encode one track as DTS w/ 24 bit depth, another as PCM audio

Holy crap... thank god I did not ask for any shorti;err; shor; err; halteri; .. anyway... I pray no-one says anything about truncate, ever again.

Yes, I won't ever mention that is (*) truncated, but -not by truncation.

* Please note that the word "is", used in this context _is_ not to be interpreted as meaning that it _is_ in fact truncated, or that truncation _is_ what in fact what has taken place, if in fact that assumption is made by a 3rd party or parties. No implication _is_ therefore expressed for the word _is_ in conjunction with any truncation that has, or has not taken place, at any time, by any party attached thereto. Your milage may vary.


[This message has been edited by soundhound (edited October 10, 2002).]

Seems in the past people talked about telephone quality versus AM radio quality versus FM radio quality versus CD quality.

And to further confuse, there's PCM versus other encodings and besides the number of bits (4, 8, 12, 16, 24) there's the sampling rate in Hz. According to my PC's sound recorder (using PCM):

8 bits at 8 kHz being is 7 kb/sec,
8 kHz at 16 bits is 15 kb/sec,
8 bits at 12 kHz is 11 kb/sec,
16 bits at 16 kHz in 31 kb/sec,
16 bits at 22 kHz is 43 kb/sec.

By the way, the source CD I used for this experiment was the newer 20 bit remastered version of "Kind of Blue", Columbia/Legacy - catalog # CK 64935, in case anybody has it, or wants to pick it up to compare.

I didn't even go there!! The CD is at the original sampling rate of 44,100hZ.

Anybody for 4 bit, 3Khz sampling rate?

This is using current AM and FM, not the just-approved digital AM and FM approved by the FCC recently: http://abcnews.go.com/sections/scitech/TechTV/techtv_digitalradio020131.html

Here's a story I related awhile back about Dolby Digital.....

"One interesting story you might be interested in regarding Dolby digital. When I am on the dubbing stage after all the final mixing is done on a feature film, and am doing my final paperwork, they are usually making the Dolby Digital master of the final film soundtrack. As I am working, I have more than a few times snaped my head up suddenly and said "what is _that_ shit!" The mixing engineers always respond: "Oh, we're playing back the Dolby Digital master to check it" I say "oh" and go on with my work........ "

In my experience, DTS is better, but not hugely so, as it still throws away data; just not as much. I don't know how many people know this, but on most DTS movie soundtracks on DVD, the bit rate is _one half_ as high as it is in regualar DTS, as on DTS CDs and in a movie theater. They do this so as not to hog the available total bit pool, and sacrifice video quality in the process.

By the way, almost all movie soundtracks are originally produced and mixed as 24 bit, 48kHZ (actually 47,952 hZ) these days. All-digital mixing consoles are becoming the norm also. On films I work on, after recording the orchestra, usually mixed directly to digital workstation, the sound never sees an analog stage until it is played back in the theatre, or on DVD at home.

[This message has been edited by soundhound (edited October 10, 2002).]

Maybe that comparison does not always apply to DPL II? As I recall, Soundhound, when you were here, you said on my particular system, you preferred DPL II to DTS at least for those tracks we compared. Although you preferred straight stereo to either DTS or DPL II, in all cases.

Will

I just got the 'Blue Man Group' DVD-Audio, (wow BTW) and digging around I notice it offers a person the chance to subjectively or objectively (if one has the tools) compare DD, DTS and the 'Master' (although not a digital master) since it has all three onboard.

Just keep in mind that the three versions might not be made from the same source (i.e. are the Dolby and DTS versions made from the analog master directly, or from the 24 bit PCM master). They might have done some subtle tweaking for the different versions in the mastering stage, in level or EQ for example. Note that at least until recently, for the DTS version of movie soundtracks, they used the 35mm 6 track magnetic film print of the soundtrack as their source. Dolby Digital masters are made on the dubbing stage, after the film is mixed, and directly from the mixing console (they actually send a guy with the 'Dolby Box" to the stage and plug him in). Not only does the DTS magnetic film version introduce it's own 'colorations" (Good ones, I'd say), but DTS used it's own proprietary low frequency equalization for it's 35 mm playback machines, boosting the low frequencies somewhat. This can, and does introduce audible differences between versions.

[This message has been edited by soundhound (edited October 10, 2002).]

Will:

DPL II can vary from DD 5.1 in it's resolution on a DVD.. The DPL could actually be a straight PCM 48k print of the soundtrack on the DVD (they might not have made the film in 5.1 in the first place, for example), which could sound better. There are many, many variables on how they can do it, and unless one has intimate knowledge of how exactly something was done, it is all speculation, and direct comparasons come down to individual preferences.

We listened to many sound segments that were not from a DVD, over a wide gamut of musical tastes, but you preferred DPL II to DTS, at least with all the sound segments we listened to, at least on my particular 950-based home theater.

Don't know if your conclusion with the sound segments you heard here, necessarily applies to other sound segments or to other home theaters.

Just curious - were you comparing PLII on a CD to DTS:NEO6 on a CD, or were you comparing PLII to a DTS CD?

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gonk -- Saloon Links | Pre/Pro Comparison Chart | 950 Review

Whew - this is getting confusing! We listened only to regular CDs. My preferences were straight stereo, then PL II, then DTS Neo 6, If I remember correctly. The preference for straight stereo is strictly a personal thing with me, especially with classical music since I'm used to having the orchestra arrayed in a particular way, with an interplay between sections happening in a way that is more 'natural' and 'continous" for me. The matrix schemes distort this balance for me; PL II is just a bit better at keeping the intent of the original imaging _for me_. Also, with popular music, especially that mastered and mixed in a ProTools (or something similar) enviornment, there are a lot of plug-ins available now that use psyco-acoustic principles to create 'virtual surround' and 'virtual imaging' where instruments can pop up anywhere around the listener, as determined by the person doing the mix. When listened to in anything other than straight stereo, most of these effects are simply steered to the surround speakers, distorting the intent of the artist and or mixer.

Of course, real surround recordings of acoustic music get around all this and capture the ambience and spaciality of the music as it should be. I just wish they would have either eliminated the .1 LFE channel, or made the LFE stereo. In my experience, having a mono sub screws up the imaging of the other speakers unless positioned and set up just-so. Even then, mono bass can't do justice to low frequency instruments like pipe organ. Oh-Oh.....there's that soapbox again (-:

[This message has been edited by soundhound (edited October 10, 2002).]

Really though I thought the theory was to allow 5 full range (bass too!) and a single dedicated LFE, at least for HT stuff. I see no big problem with this as long as the recording engineers don't assume a 5 sat/1 sub system and downmix or limit the '5' for us at mixing time. I always thought of the .1 as a 'boom' channel for HT, and also in systems where any of the '5' are not bass happy it is a possible place to steer that excess baggage, but I would hate to think someone is intentionally limiting the '5' part 'at the source'.

I believe you are right on, charlie. This is/was the intent of Dolby Labs.

Charlie:

I'm not quite sure I follow your drift, but anyway... I think having 5 full range speakers is the way to go for SACD and DVD audio in multichannel, and having a "boom channel" for movies is the way to go, also....but.....It's just that multichannel music discs are such a new medium, and there's no real standards on how to devide up the bass. I think that the .1 LFE is pretty useless for music (personally) and that all the speakers should be full range (the .1 not used for anything). But most people have smaller satellites, and therefore that .1 raises it's ugly head, and the temptation to use it by the mixers of the music discs, forcing their bass management on you. What to put there?? That's the problem. I just think that with a mono subwoofer, there's just no elegant way to do it (I'm talking about music discs only here) that does not mess with the imaging. I myself don't make use of the LFE at all, even in movies (you might know this already from previous threads). I re-direct the LFE and all the bass from the other speakers to the front left and right mains, and the bass is re-directed by an electronic crossover at 60hZ to stereo subs which sit next to their respective left and right main speakers. I guess my point in all this is that I'm approximating full range speakers all round, while keeping stereo bass ambience, and still getting the punch from the LFE track in movies. Geez - I think I'm rambling on again...........

[This message has been edited by soundhound (edited October 10, 2002).]

soundhound: i am confused. first, you say that you do not make use of the lfe. then you say you re-direct the lfe through your mains and low pass it to stereo subs.

if that is correct, you DO make use of the lfe, and your setup would be one that is preferred by some. i personally see no difference in this setup vs. the standard setup of lfe plus under 80hz summed and sent to the sub other than your belief that sub 60hz has ambiance in stereo vs. a good mono sub.

the question i have is simple. exactly how do you "re-direct the lfe to the mains"?

I seem to be taken to task for just about anything I say lately! Let's see if I can make sense after ingesting a pitcher of Margaritas..... (-:

I do not use the LFE output of my processor. Instead, I direct all bass, including the LFE to my main left and right front speakers. In line with them I have an analog 60 Hz crossover that feeds that bass to a 1000 watt stereo amplifier, then to two 18" sub woofers per side, in stereo. I mainly have it this way because I have three possible signal paths that must work togeather seamlessly. 1) a home theatre path from the 950 and my Sony EP9ES DD pre/pro. 2) a direct path injected at my tube preamp for 'purist' listening in stereo, that goes to the main left and right speakers alone, but where I still want to use the subwoofers, and 3) a path I use when I work, directly from my mixing console where I usually monitor in 6.0 (less LFE) so that I can hear how things will sound when finally mixed on the dubbing stage.

If I am still not making sense, maybe we can both go out to this great Mexican restaurant I know, and have another pitcher of Margaritas (or your favorite beverage) and try again; sometimes it helps.....Well, yes, it _always_ helps (-:

[This message has been edited by soundhound (edited October 11, 2002).]

SH: i appreciate the reply. i am certainly not taking you to task. i understand your setup from earlier posts. i only do not understand how you redirect the lfe output of the 950 into your mains.

i find nothing on that in the manual, and would like to try it in my own system.

My setup in the 950 is: Front main left and right are set to "large". Surrounds (side and back) set to "small", with a crossover set to 60hz. Center set to"small", crossover set to 150hZ. Subwoofer: set to "none". The 950 then directs _all_ the bass, including the LFE to the front left and right mains, where it can be crossed over by analog to your subs. BTW, I set the center crossover so high because I like a bit of 'leakage' from the center to the left and right at low frequencies: it kind of 'spreads out' the dialogue across the soundstage, and helps, at least in my setup, to take the edge off the 'mono-ness' of the center speaker.

How about using KaZaA as the transport medium, then we could download the file from multiple sources? I would be willing to start the process. My KaZaA ID is DSLine@KaZaA

[This message has been edited by MeanGene (edited October 11, 2002).]

shound,

From reading some of the threads on other forums about the variable crossover settings I'm confused a little about the bass management on the 950 in a setup like yours. The crossover settings obviously set a high-pass filter for the speakers, but does each individually also create a low-pass for the summed signal to the sub, or is a full range summed signal from each small speaker sent to the sub with a single low pass point?


brianca

Since my surrounds have a crossover point of 60hZ, and my center has one at 150hZ, there would have to be a seperate low pass generated for each, which is then sent to it's respective channel. i.e. left surround bass below 60 hZ is sent to the left front, right surround sent to the right front, and the center bass below 150hZ is split, and sent both to the left and right fronts. At least that is _supposed_ to be the way it works! In my amp rack, the left and right channels have a stereo analog 60hz crossover network before the power amps that sends all the bass below 60hZ to the sub-woofer stereo amplifier, and a high pass that sends everything above 60hZ to the front mains.

In the 950, even if all the channels have the same crossover point, there is still a low pass generated at that frequency which is sent either to the speakers set to 'large' (always the front left and right, but I don't imagine this has to be) if no subwoofer is selected, or to the LFE output if there is a subwoofer present (as set in the speaker setup menu).

I'm just realizing that this is one of those things that is easier for me to visualize, than to try to explain in writing to someone else, so I hope I've answered your question.

[This message has been edited by soundhound (edited October 11, 2002).]

Gene:
I've not heard of KaZaA. Can it handle a 95 Megabyte file? Let me know if it's something that I could simply upload to or how it would work.

The CDs should be getting to the people I sent them to maybe tomorrow (Friday), or at the latest on Monday. There are a couple who are going to host it on their servers, but another source would be great.

I'll drop it on kazaa as well when I get it.


brianca..

shound,

Here's one of the articles I was talking about.

http://www.hometheaterhifi.com/volume_9_3/feature-article-multiple-crossovers-9-2002. html

What's your take on this?


brianca..

Brianca:

Very interesting article; thanks for the link. This all points up my main objection to the current 'bass management' thing - it's confusing as bloody hell to just about anybody, and when you factor in SACD and DVD-A, it's worse! I'd bet there are _very_ few people out there who can make sense of all this stuff: it doesn't pass the "my grandmother can do it" test!

He's quite right that when summing signals to be low passed, you do the summing first, then low pass the result. Phase anomilies result otherwise. That's one of the reaons I do it the way I do, (not using the LFE output of the processor: setting it to "no subwoofer"), avoiding the whole mess. I noticed that he described my setup exactly in the last sentence of the article, too. In an actual film dubbing stage, they don't use 'bass management'. The LFE is just sent directly to the sub woofer as a seperate channel. Setting the low pass on a subwoofer (in a conventional setup) really low, like to 40hZ is not really much of a liability in my opinion. Any 'hole' that results may not be audible as such. Things get pretty messy in the bass region below 100hz on any system, in any room, so differences might be swamped by such things as standing waves in the room etc. Short of using a real time analyzer and calibrated microphone, it's going to be a comprimise in any system. My recommendation, tune it by ear if that's the only option, and enjoy the show!

I _am_ taking it as an article of faith that when I set my surround crossovers in the 950 to 60hZ and the center to 150hZ that it does what I would expect it to do as I outlined in my original post. Now you've got me thinking. I'll have to measure exactly what's going on to confirm. Luckily, my surrounds and center are actually capable of running full range. In the 'mixing console' configuration of my system, they are in fact receiving a full range signal.

Thanks for bringing all this to my attention!

[This message has been edited by soundhound (edited October 11, 2002).]

4 years ago when only Theta processors had adjustable crossovers, I experimented with global xovers on other processors at 100Hz, 90Hz, and 80Hz and found they just didn't work well with my Dynaudio speakers in my room.

Therefore, in my processor's setup menu I set the Sub=No, Mains=large, center and surrounds=small on the processor and added an electronic xover (@60Hz) between my main towers (Dynaudio Contour 2.8) and my powered sub.

This essentially sends all bass from the small channels, plus LFE, to the main L&R channels.

This has allowed me to use the sub for both 2-channel music and HT. I hear those standup bass string plucks and HT explosions just the way I want without having to adjust the sub output constantly.

I also use a parametric EQ on the sub and sophisticated acoustic software to tune the room's bass.

The electronic xover has level controls for low-pass, high-pass, and xover frequency and uses 24dB slope Linkwitz-Riley symmetrical filters.

bstan:

Yep, your setup is pretty identical to mine, and you find exactly the benefits I've found - it is very 'set it and forget it' no matter what I shove at it. The only wrinkle in mine is that my subs aren't powered so I have an outboard amp, and I run stereo subs.

I really wish the industry went the external crossover route (or they could integrate it into the pre/pro) in the first place - it just seems so much more logical and beneficial, as least to my twisted brain.

[This message has been edited by soundhound (edited October 11, 2002).]

For anyone interested, I'm uploading a wav to my site ( this page ). It'll be another 60 minutes or so before it's done, but I'm off to dinner now. Haven't even had a chance to listen to it yet! Enjoy...

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gonk -- Saloon Links | Pre/Pro Comparison Chart | 950 Review

[This message has been edited by gonk (edited October 11, 2002).]

I just got back from dinner, and it's been up for a little over 10 minutes. Go get it!

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gonk -- Saloon Links | Pre/Pro Comparison Chart | 950 Review

Sure I have DL'd much larger than that. Make sure to get KaZaA Lite, so you don't have to deal with the Advertisements. You can get the KaZaA program at: KaZaALite

MeanGene:

It must be a PC program; I use a Macintosh. Guess that's what I get for 'thinking different'.

V

Are you sure about that?? I think I got about five pop-up advertisements as soon as I went to their website.

Here's what I got from the manual on the crossover settings:



So it says that the crossover settings set the high pass for the speaker groups, but not that it also sets the low pass for the signal sent to the sub. It doesn't say anything to imply that this isn't done either, so maybe the outlaws and step in here and clear it up.


brianca

[This message has been edited by brianca (edited October 12, 2002).]

[This message has been edited by brianca (edited October 12, 2002).]

every tower speaker, center channel speaker and surround speaker worth a damn has a passive 2 , 3 or 4-way crossover built in.

the only way high pass filtering the signal to any of those speakers before the amp makes sense, is if they are a single speaker that isn't meant to receive a full range signal (ie, live sound setup, bi-amping or tri-amping).

therefore...all speakers should be set to LARGE.

that leaves the lfe channel. i have yet to read one single word that details the 950's dsp "precision bass management" feature, especially, what happens to the lfe signal with sub set to 'no' and with the high pass filters set to any choice.

someone point me to the correct page, please.

if i assume what has been posted in this thread to be actually the answer to my question (sub set to 'no', lfe is digitally 'y-jacked' and sent equally to L/R mains along with any info from channels set to SMALL below whatever high pass point may have been selected), then:

a.) your sub, or subs are receiving double bass in all formats.
b.) your mains are receiving less than a full range signal, even though they are set to LARGE.
c.) you have phase problems, even if all SMALL speakers are set to the same crossover point.
d.) your setup is less than setting the sub to 'yes', as you would have full range to the mains that way, otherwise it is no different than the 'standard' setup.

bass management is actually very simple...

a.) set all speakers to LARGE.
b.) buy a sub with an onboard bypassable preamp, adjustable phase control, parametric eq (optimized for lfe) and variable low pass filter.
c.) connect the analog lfe output of your DVD player directly to the sub.
d.) set the sub to 'yes' (nothing is connected to the prepro's .1 output)

no double bass, no phase problems that can't be easily adjusted on the sub's front panel, good sound for HT or any music format.

am i crazy? stop by and listen. i've painstakingly tried all three setups with every combination of crossover setting. this setup works....and grandma handles it very easily.
soundhound...you are a bad influence...i am not used to margaritas at all, much less a pitcher full! though,it sounded like a good idea at the time.

I'm sorry but that's not always true. Many people, including me, have speakers that play low frequencies but not loudly, without distorting, awfully. The same speakers play higher frequencies loudly without distorting.

I think you would agree that for these speakers, I should use high pass filtering, rather than hear distortion (and risk speaker damage).

Will

[This message has been edited by Will (edited October 12, 2002).]

will: i do agree that the so called bass management is for people who have small speakers. these are usually the surround speakers and the center channel speaker.

unfortunately, the bass is not managed very well at all and the confusion as to what actually happens to the bass is apparent by reading this thread or any article on the subject.

all 5 speakers should be identical and full range. short of that, the thx spec of setting all 5 speakers to small and xover at 80 for all 5 is the surest option, though it has the incurable double bass problem.

mixing large and small speakers leads to less than optimal listening.

if you have a 950, a good sub or subs, 5 or 7 channel amp, 1080i monitor, good dvd player with sacd or dvd-a and pt barnum interconnects and cables....why would anyone let the speaker setup compromise the best performance possible?

Bosso:

Sounds like the setup you describe will handle it all, and very handily. My grandmother might have a bit of trouble setting it all up though, your grandmother's milage will obviously vary.

I don't know however how you get the conclusions of double bass, phase problems, etc. in my particular setup. The surrounds send any bass below 60hz _supposedly_ to the L&R fronts. The LFE is "Y"ed to the L&R fronts, and the center's bass is likewise "Y"ed to the L&R fronts. Now, from recent posts about the possible workings of the 950s 'bass management', I could be totally, and completely and utterly _wrong_ about my faith in the steering being what I assume it is. The only way this could be checked out for sure is for me to buy that Dolby Digital mastering software for my workstation so that I could make a test DVD. I've been putting that off since it's expensive, and I've not had a huge demand for what it does. Maybe that time is coming.

All my speakers are very capable of running 'large'. I am seriously going to re-evaluate running them that way, and only running the LFE to the main L&R for distribution to the subwoofers by the external crossover. Actually and truthfully, the only reason I started running the surrounds in 'small' is because I was driving them so hard during the storm opening scene in "Twister" that the woofers hit bottom with a big 'snap' - I'm sure you know that sound! The reason I started doing it for the center is because it sounded a bit tubby, since it sits on the floor (it has two 12" woofers).

I have to admit that I don't _hear_ anything unusual that would suggest double bass/phase problems etc. when listening to DVDs, but I might just change them back to 'large' after all. I do most of the really critical listening to my system with the 'mixing console' path, so when I listen to a DVD, I kind of lower my critical standard. And besides, in the mixing console path, I feed full range signals to all the speakers anyway, since I don't use any bass management on anything other than the 950.

Anyway, Tweak time!

[This message has been edited by soundhound (edited October 12, 2002).]

Bosso, since money's an issue for many people including me, and since the front speakers usually have more sound and dynamics than the surround speakers, a case can be made for using better speakers in front and lesser speakers as surrounds. It's cheaper than using identical large speakers, all around. And should sound better in general than using small speakers all around. Using small speakers all around would not be a better solution, unless there's as much sound and dynamic range from the surround speakers as is from the front speakers. I'd love to get full range speakers all around, but can't afford it. Fortunately unlike Iggy the Dog, I don't have sensitive dog-ears.


[This message has been edited by Will (edited October 12, 2002).]

The new $3500 Anthem AVM-20 v.2 THX ULTRA2 programming recommends that you simply set your global crossover to the THX rec of 80HZ. Last I checked... the Outlaw unit allowed that as well.

[This message has been edited by merc (edited October 13, 2002).]

SH: you hit the nail on the head earlier...it really IS harder to write it in logical order than it is to visualize.

i'll try, nevertheless.

bass below crossover (60hz in your case), summed with lfe (bass below 120hz) and bass to mains (as low as the program), equals double bass. if you low pass that combined signal to your subs at 60hz, your subs get the bass info below 60hz from the L, R, SL, SR, center and lfe signals.

there is most certainly a phase issue there, but, if you don't hear one, that isn't a big problem (though i can't imagine what 2 18's driven by 1,000 watts sounds like either way...and, by the way, on that subject...too cool). the main problem is that you have double bass in the subs. it ain't a bad thing either, but you have trained ears...you would hear a definite difference with full range to the mains and only lfe in the subs.

and btw, you don't have dbl bass in stereo with your setup, in fact you have the ultimate stereo setup....must sound very good.

I'm missing something here. The defintion of a crossover is that it high passes the mains (center, surrounds, and rear(s)), and low passes the sub.

Any time alignment should happen *after* the summing and crossing over. (That just makes engineering sense.) (The 5.1 analog inputs are a difderent story.)

I have spent a nontrivial amount of time making sure with discrete tones that my mains are in phase with my sub.

Am I missing something here?

bosso...



Actually your statement above isn't true. You might be quite surprised how you can improve both bass frequency accuracy in the room and the mid-bass to mid-range clarity of your main speakers when you high-pass tower speakers. You actually reduce the THD of the tower's bass/mid-bass driver.

You also said;



Remember, there is an active xover splitting all the bass going to the main channels between the mains and the sub(s). So your point (a) is false, point (b) is obvious, point (c) is false.

In your setup as you specified above, if the lfe is decoded from the DVD player and the rest of the channels are decoded from the processor, do you think they are actually timed correctly? I doubt it.

Have you used a professional MLS based acoustic measurement system and observed the waterfall charts in the bass region? Note any bass frequency timing anomolies? With your setup I would expect to see timing problems between the DVD and processor DSP outputs.




[This message has been edited by bstan (edited October 12, 2002).]

bstan:
a vented enclosure will benefit with a 2nd order high pass filter before the amp, but...the high pass frequency and the Q must be very specific to the driver/enclosure, otherwise, you'll get a huge bump or hole (depending on the Q of the high pass filter, which is not published by any prepro man. that i know of, and therefore not predictable) at crossover point. give me your driver parameters, enclosure volume and vent specs and i'll e-mail you the graphs at all crossover points available on any prepro.

i am aware that there is an active crossover "splitting ALL THE BASS going to the main channels between the mains and the sub(s). to make it simple, take a low E bass guitar signal (approx. 42hz.) that is mixed into the lfe channel. simultaneously, there is the same key struck on a piano that is mixed into the L&R surrounds while a midi kick drum of approx. the same frequency is mixed into the mains. all three signals are now summed and directed into the sub(s) slipping under the 60hz. low pass point....double bass (actually, triple bass). not to mention, the producer's meticulous efforts are negated.
some of the harmonics of those 3 occurrences of a 42hz note remain where they are supposed to be (in this case, above 60hz) and may give the illusion of being correctly placed, but the whole idea of discreet music formats is null and the bass is muddied, exaggerated at the crossover point and out of phase.

the phase problems come from the summing of the bass signal of channels with different high pass points (ie, center at 100, surrounds at 60, mains at 40) with the lfe signal. read the article brianca provided the link to above.

as to the timing problem in my setup, it is not so easily, yet definitely corrected by a delay adjustment at the sub's crossover. (remember...the same delay problem exists within the design of most main speakers and is not adjustable, but that's another discussion in itself)._



[This message has been edited by bossobass (edited October 12, 2002).]

KCB,

High Pass crossovers and Low Pass crossover are completely seperate. To say crossover does not mean both. The sub is a good example where most setups have a low pass for the sub and few have high pass filters on that signal. The 5.1 analog crossover on the 950 is another example where the switch only activates a high pass to the mains but does not enable a low pass for the sub ( at least I think that's right).

You can't assume that when you say crossover you mean both.


brianca..

I suppose this bass management and redirection is only an acedemic debate. If not, your vexations over technical aspects must surely impede the enjoyment of listening.
For what it is worth. I feel almost any speaker out there (that is affordable by most and some) will be greatly enhanced by a seperate subwoofer/woofers. Plus most systems will function better when the massive power requirements of the sub are self dedicated. I recently added a 15" Tempest to my system with the 1050. My speakers when measured by a SPL meter rolled off at about 40Hz. After many measurements I found the flattest response (down to 20Hz) I got with intergrating the sub with my speakers was setting the crossover to 100Hz. Technically I should have been able to crossover lower; probably at 60Hz but the 100Hz gave me the flattest response (used the SPL meter correction charts also). Why at 100Hz I don't know. Room acoustics, accuracy or the functioning of the crossover in the 1050? Beats me. The sub is driven by a regular amp (no crossover).
Now that it is set I sit back and enjoy my music. But then every now and then a dark cloud crosses my mind; would some Burr/Brown DACs sound better than the AKM DACs of the 1050? Hmmmmm.

The key is to experiment with the crossovers and see what works best.

In my experimention, I had the exact opposite experience as you. Initially I crossed my main speakers at 80 Hz thinking, when in doubt, use a slightly higher crossover than necessary. But I subsequently experiemented with 60 Hz and found the sound integrated better, so kept it at 60 Hz. Later still, I experimented with 40 Hz and found the sound to be even better. So now my mains are crossed at 40, and my rears and center are crossed at higher frequencies.

Actually, I find I like my sub/sat setup more and more in stereo + sub mode, without the center and without the rears. For imaging, the 40 Hz is the way to go, with my stereo + sub setup, on the 950. My mains are Von Schweikert VR 4 Gen II's. It's specs say its -3dB point is 20 Hz, fwiw. BTW, I had to fiddle with phase of subwoofer when changing between 2 channel + sub, to 5 channel + sub.

Originally posted by MeanGene:

Quote:

Make sure to get KaZaA Lite, so you don't have to deal with the Advertisements. You can get the KaZaA program at: KaZaALite

Originally posted by Will:
Are you sure about that?? I think I got about five pop-up advertisements as soon as I went to their website.

I meant from the KaZaA program not the web site. I am going to host the file on my computer if ya want it. You should be able to find the file by searching for "soundhound".

By the way SoundHound, What is the best frequency to shake the margarita mix with? 20Hz? I have a pitcher sitting on the sub now.


[This message has been edited by MeanGene (edited October 12, 2002).]

I am naive about p2p programs but I'm worried about what it does when it's executing on our computer. And are you sure it doesn't cause spam and that it doesn't do a trojan horse virusy number when it's on our computer? Again, sorry to ask what may be a real naive question, but I really really want to be sure, before downloading it...

Bosso:

Thanks for the 'splaination. Yeah, I am going to resotre tha 950's to all large. Easy to do, and the speakers can take it - I'll just have to moderate my demos of "Twister"! I was playing it _way_ louder than reference level at the time, _Waaaaay_ louder.

Acutally, it's two 18" subs _per side_, four total.

The lite version has most of the spam and spyware removed. I also use anti-virus software, a software firewall, a hardware firewall (router), and use the Adaware software to remove or opt-out of things that are using my computer without my permission. You don't need these tools, but I wanted you to know that I have used them to check on software like KaZaA and found the lite version to be accetable as far as P2P is concerned. Never the less, if your on the internet you should have virus protection at all times, even if your just surfing. I also have a PopUp Stopper which is why I didn't notice the problem with the KaZaA Lite web site.

MeanGene:

Well, I've found that 27hZ is the ideal frequency to get a good, frothy Margarita. Of course that depends on getting _exactly_ the right amount of ice and Triple Sec. But after a pitcher, hell...I don't think it matters much------_Nothing_ matters much

BTW: stay off that 'sycle after the above...


[This message has been edited by soundhound (edited October 12, 2002).]

What is the advertisement problem with the non-lite version of KaZaA, and might those problems easily appear on the Lite version, as well?

The adverts are on the official web site for KaZaA Lite. It appears heavily advertized funded. Would we see advertisements and perhaps even popups (if you did NOT have a PopUp Stopper) after downloading the latest version of the KaZaA Lite to your computer?

Again, I apologize if these are naive questions....

[This message has been edited by Will (edited October 12, 2002).]

Onto the original topic - Has anybody had a chance to listen to the "listening challenge" CD yet???

Ok.. It's on Kazaa now. File names are shound-challenge.wav (97MB) and shound-challenge.zip (67MB). My username is vic_badger@kazaa.

I'll get it on a web site before the night is over, and I'm headed in to listen to it now.


brianca.

KaZaA is a p2p program that does have alot of advertisements in the program itself. KaZaALite is a hacked version of KaZaA where the program has been modified to remove the ad stuff, except for the first welcome screen that asks for donations. The unmodified version has ads running all the time in the application, poping up in your face and using your bandwidth.

If KaZaALite starts working like the KaZaA version I will uninstall it and start looking for something else. You have to remember that this is unaccetable by some of the people that create music. Your getting Music for free. I contend that the Music that your getting is not the same quality and there for is a kind of demo of the Music. I listen to it and if I like it I will go out and buy the CD. Most of the Music is in 128 bit/sec mp3's with CD audio quality being 320 bit/sec quality. KaZaA has generated many sales, like the Blue Man group audio DVDA that I recently ordered.

I hope that answers your questions.

Also posted at website:

http://65.112.245.88

brianca

So, is that article that brianca posted correct? That the upper range of the LFE is limited by the lowest common denominator crossover of the other speakers that are set to small? How would I test that?

Hell, who knows!?!?!?

This is _exactly_ why I hate the "bass mis-management" scheme that's in all HT gear. Nobody _including ME_ can really wrap their minds around what the manufacturers are exactly doing - and they're no help in clearing it up. I would take the article at face value, as he at least is making sense, and sounds like he's 'sessed it out.

The whole problem in my twisted view, and coming from the industry where this whole LFE mess started (the movie industry) is that they tried to force the actual dubbing stage and movie theatre model on a home envoronment. But _they_ don't use bass management anyway! It's all straight full range speakers, except for the sub. The LFE channel is a decendent of the 'baby booms' from the days when there was a transistion from 6 track magnetic soundtracks in common use and five channels behind the screen. They started using the half-left and halr-right speakers as 'boom channel' (circa 1977 'Star Wars"). This led to the LFE we have today, and only three speakers behind the screen. The problem arises on how do you integrate a LFE channel into a system in a home environment. The big mistake I beleve they made was not mandating that _all_ speakers be full range in the first place. Then no bass management, and no problems. Simple. Then they got the idea that most people wanted smaller speakers. Personally, I think they just thought that the costs of speakers would be too prohibitive if they were all capable of real bass. I disagree, but anyway, thus was born bass-management. Now, I have articles from "High Fidelity" magazine from the 1950s (I'm not _that_ old, however!) that detail a sub/sat speaker system in mono, then later when stereo started, in stereo. The idea is not new. But they didn't use any form of bass management other than a plain 'ol analog high and low pass crossover to devide to the sub.

I think you might get the impression that I hate bass management


[This message has been edited by soundhound (edited October 12, 2002).]

There's something missing from this thread. I know there were a few other posts that were here last night, and I asked a question too, that aren't here now... About bass management. Strange.

Yeah, pg 2 and pg 3, duh!


[This message has been edited by Kevin C Brown (edited October 13, 2002).]

You mean aside from the crossover question that you asked about halfway down page two?

brianca.

[QUOTE]Originally posted by bstan:
[B]bosso...

Remember, there is an active xover splitting all the bass going to the main channels between the mains and the sub(s). So your point (a) is false

bstan: you are correct, sir. there is no double bass in this setup, the bass is redirected. i momentarily confused this setup with the thx setup when any of the 5 speakers is set to small and low pass is set above the selected high pass for the small set speaker(s).

my humble apologies and thanx for catching my mistake.

bosso,

No problem. As I said and generally agree, bass management in the processor has been questionable from the beginning, which is why I did most of my bass management between the L&R mains and sub(s) over 4 years ago and I'm still doing it that way today.

Simply sounds and measures better for me, with the benefit of working in both 2-channel and HT. I have a stereo preamp as my control center and route the processor's L&R outputs to it.

[This message has been edited by bstan (edited October 12, 2002).]

Do you really mean this literally? It is generally agreed by most all audiophiles that if your speakers are not capable of reproducing the frequency range from 20kHz down to 20Hz -no more than -3dB down- then you do not have full range speakers. Period. Those speakers that are capable of this are very expensive- I bet not many participating at this site have such a set-up. (Especially 5 of them). I do not consider a speaker/subwoofer setup using an external crossover to meet the criteria of a full range speaker, even in a stereo arrangement. As charlie said the other day, Dolby 5.1 was designed for 5 full range speakers and a subwoofer for LFE. (If, as you say, the studios choose to ignore the LFE channel, then I suppose that is their perogative). If "they" had mandated full range speakers only, most of us would be looking for a new hobby. IMHO, bass management_for most of us_using our pre/pro's capabilities works fine. I have no problem with your setup- in fact there are some advantages to be had with it. I just think the intention was/is to have an _easy_ and workable solution for the "masses". By the way, I have 2 subs that I used to operate in similar stereo configuration to what you have going (using an external stereo crossover). I now have them stacked near a corner and get better results in my room for both HT and music. Not perfect by any means, but I'm still working with them. YMMV! And then you have **THX**! So, everybody -- set all your speakers to "small" and crossovers at 80Hz and be amazed at how well it works! But don't tell anybody 'cause most will never admit they have anything that's considered small.

[This message has been edited by steves (edited October 12, 2002).]

I downloaded the .zip file and unzipped it into the .wav file. I sent the .wav file to the computer that has the plextor cd burner and burned a cd (burner set to slowest burn speed of 4x). I also sent the .wav file to another computer that sits next to the stereo. I will hook this computer up to the stereo and play the .wav through via the turle beach santa cruz sound card. I will also play the cd through the same stereo and compare the .wav file sound via the computer to the burned cd playing in the audio cd player.

I will have to listen to the file a few times to become familiar with it before I listen with the purpose of detecting when the bit depth changes between 16, 12 and 8 bits.

If I can't hear any changes, then I'll certainly have one severe case of audiophile letdown. I will need years of therapy, perhaps in Tahiti. But on the bright side, I can sell all my audio stuff and just use a kenner close-and-play. Maybe they have an updated one that plays DVDs.

------------------
the 1derful1

Steves:

I never said that movie sound mixing stages ignore the LFE channel - Just that bass management isn't used anywhere except in home theatre. Please re-read what I said above.

[This message has been edited by soundhound (edited October 13, 2002).]

After you give it a try, post your impressions here, then Email me at soundhound@earthlink.net and I'll send you the 'cheat sheet" that has the timings of all the changes.



[This message has been edited by soundhound (edited October 13, 2002).]

[QUOTE]Originally posted by steves:
It is generally agreed by most all audiophiles that if your speakers are not capable of reproducing the frequency range from 20kHz down to 20Hz -no more than -3dB down- then you do not have full range speakers. Period.
____________________________________________

i don't know much about 'most audiophiles' opinions, but there is virtually NO program fed to the 5 main speakers below 40hz., multichannel music or HT soundtrack. this qualifies a lot of affordable floor standers.
i have found that there simply aren't many 'audiophiles' comments available in the multichannel digital format arena. i personally consider many of the people who post here to be very knowledgeable in this relatively new format.
___________________________________________

I do not consider a speaker/subwoofer setup using an external crossover to meet the criteria of a full range speaker, even in a stereo arrangement.
___________________________________________

i don't understand this comment.
_____________________________________________


And then you have **THX**! So, everybody -- set all your speakers to "small" and crossovers at 80Hz and be amazed at how well it works!
___________________________________________

i have modeled every speaker in my database and found that implementing a high pass filter before the amp (at 40, 60, 80 and 100hz.) causes a huge 6db hump at the crossover point. in fact, the resulting strain on the driver causes it to unload in the vast majority of cases at reference level. if you select 80hz as a high pass to all 5 speakers, and if lower than 80hz as a lowpass on your sub sounds better to you, you lose information. if higher than 80hz sounds better as a lowpass, you have double bass. soundhound, bstan, et al, have eliminated theses shortcomings with the redirection of the bass info, but still have changed the original mix of many multichannel programs. how can this ever be considered 'audiophile'?

both the producers of surround soundtracks (see soundhound's comments) and multichannel music discs (see the telarc link) use 5 full range speakers and a sub with no bass management to monitor the mix of the the masters. it only makes sense to play it back the same way...masses be damned.

Bosso:

You're right that there's usually no deep bass in the main speakers of a film soundtrack. I know - I sit right there at the mixing console when the films are mixed. The main front speakers on a dubbing stage are sped'd down to 30hZ, the surrounds to 40hZ. All the real low bass goes to the speaker that is built to take it - the subwoofer. That's where that little track on my workstation's screen labelled 'LFE" goes...........

I wonder what your take is on this: what ever happened to 'acoustic suspension' speakers in the transistion to Home Theatre? Those speakers were certainly capable of going down to 30-40hZ easily and cheaply, and with modest cabinet dimensions. Sure they were inefficient, but amplifier power is relatively cheap. Can you think of any reason why they would not have been a better solution than a vented box, which can flail it's woofer if presented the wrong frequency and level of bass?


[This message has been edited by soundhound (edited October 13, 2002).]

True. There is not much information below 40Hz on most music recordings either, but that does not change the accepted definition of what constitutes a full range speaker.
Interesting! I would add that_generally speaking_ speaker internal crossovers can- and do- cause similar humps and/or dips in some models available.
Absolutely true!
Agreed. If you have 5 full range speakers. Yeah, to hell those masses anyway!

bosso:



This is one reason I use the symmetrical L/R -24dB slope for my low-pass and high-pass filters to the sub(s), it sums to zero dB gain and zero degree phase angle offset.

Most ported main speakers are probably in the -18dB to -36dB acoustic rolloff category all by themselves, unlike the well behaved 12dB rolloff of sealed acoustic suspension speakers.

Add this excessive rolloff of the ported speaker to the electical -12dB high-pass rolloff of the processor and you get crap for a smooth bass management xover. Who knows what kinhd of phase offsets in produces.

Just some of the reasons bass management is such a rats nest.

One of the inherent problems of a sub satelite system is frequency humps and frequency holes.

It's my understanding that frequency humps are more noticable than frequency holes, and for that reason, designers may tend to not want to make the high pass crossover and low pass crossover exactly the same, since that might cause a frequency hump.

Better to have designed-in fequency hole instead than risk having a frequency hump.

A sub-sat system with a hole sounds pleasing, until compared against a full range system. The rull range system sounds better.

Take the wife and kids and run for the hills - film industry secret plot revealed! When, in a film mix a source is double assigned to both the mains and the LFE (very common) - the result is.....The result iiiiSSSSS_______DOUBLE BASS!!!!!!!!!! YEhaaaaaaaa!!!! Oh Nooooooooo.........


[This message has been edited by soundhound (edited October 13, 2002).]

Huh? When I went through the process of eq'ing my sub, I plotted my sub alone, my mains alone, and the sub + mains, from 20 Hz to 98 Hz (Autosound 2000 test CD) through the 60 Hz crossover point. I never saw a 6 dB rise. In fact, most crossovers I'm familiar with, set the high pass filter for the mains at -3 dB and the low pass filter for the sub at - 3dB at the crossover freq so that you get *flat* freq response through the crossover point...

[QUOTE]Originally posted by soundhound:
[B]Bosso:

I wonder what your take is on this: what ever happened to 'acoustic suspension' speakers in the transistion to Home Theatre? Those speakers were certainly capable of going down to 30-40hZ easily and cheaply, and with modest cabinet dimensions.
_________________________________

acoustic suspension, or sealed box speakers, are everything you say except the 'cheaply' part. the driver has a low Fs due to the greater cone mass required, super-high excursion, a large magnet structure, long voice coil and a cone that won't distort
when subjected to the radical forces necessary at low freqs. it costs a pile-o-dough to build a good one.
_________________________
Can you think of any reason why they would not have been a better solution than a vented box, which can flail it's woofer if presented the wrong frequency and level of bass?
__________________________

vented enclosures aid the driver to acheive a response lower than Fs. and, because the vent takes over at the lower freqs, the driver is less taxed. it is usually more efficient with less distortion due to less excursion...and...is cheaper to build. yes, it can unload at lower freqs, but this is easily overcome by inserting...yes...a high pass filter before the amp. (the filter Fx and Qx must be specific to the driver, is the catch). bottom line is, vented drivers/enclosures are cheaper to produce with better results. the car audio people use acoustic suspension more often because most vehicle interiors have an acoustic response that rises at 12db/ octave from about 50hz, giving the sealed box (12db/octave roll off, as bstan said) a flat response down low. (plus, they seem to have unlimited funds for drivers, caps, batteries, alternators, amps, etc.)

One big problem is deciding what to measure when looking for or trying to avoid cross-over 'humps'. Do you want constant voltage, constant power, etc.

One would want constant spl (sound pressure level), which would correspond to constant power (assuming reasonably flat speaker frequency response). I remember years ago a discussion about active crossover design. One design used active circuitry to get a high pass or low pass signal that was used for the appropriate driver(s). This same signal was subtracted from the full range signal to get the complimentary low or high pass signal. Example: take a full range signal and using Rs and Cs and whatever active device of choice, get, for example, a high pass filter which can be used for a tweeter or a low pass limited speaker system. Take this same low pass limited signal and subtract it from the full range signal and you have a (derived) low pass signal which can be use appropriately.

I recall the discussion of this derived ciruit: the general consensus was that when the active signal was summed with the derived signal, proper voltage was the result. One wants proper (flat) power, which this derived approach does not give.

I have my ICBM between my power amps and pre amp. The only bass management done in my system is done by the icbm is it is used to send the bass below 60 or 80 Hz (I am still playing around with it) to the subs.

Paul

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the 1derful1

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Questions:
1.) your sub+mains were flat from 20-98hz?
2.) did you move the mic around the room for each plot point?
3.) is the room acoustically corrected?
4.) did you use a 1/3 octave real time analyzer, or a rs spl meter?

points:
1.) we're talking about a low frequency high pass filter, not a crossover network.
2.) depending on the Q of the filter, which to me at least is unknown, a hump or a hole may be the case, but NEVER flat when the filter frequency is above Fs.
3.) my software contains only info for 2nd order active high pass filters. i am unfamiliar with digital filters (and would like any info or links that might clue me)
4.) the 950 specifications list only analog filter info, i can find not one word about 'dsp precision bass management' in the manual...i would also appreciate any info anyone may have on this subject.
5.) i am basically clueless to any test or plot anyone may have made on there setup at home. i only know what my software tells me, and it virtually never lies. i would surely like to know the answer to this part of the bass management puzzle.
6.) in the meanwhile, i do not use bass management at all. i plays it da way dey mixes it. also, if i fail soundhound's 12bit vs 16 bit test, i'm gonna listen to only AM radio for the rest of my life, and all of this will be moot. (i think he rigged the disc)

Well, I've listened to the sound challenge .wav file a few times now.

I have not been much of a jazz fan, but I enjoyed this music very much. Maybe it is time to get a new/better CD player, one with SACD capability perhaps and to start exploring jazz. Then I will listen again and see, oops, hear what I can hear.

Some general comments/questions.

Was this taken from a vinyl record?

I ask this because at times, especially on trumpet "blasts", I can hear what sounds like to me as either record damage or cartridge mistracking. Also, in the very last few seconds, as the music is fading out, I can hear what sounds like to me as line induced hum (60Hz or 120Hz hum). Bings I do not know when this recording was recorded, I have no idea what limitations the recording equipment.

I have the most confidence in identifying the transitions from 8 bits to 16 bits and almost as confident when going from 16 bits to 8 bits. 8 bit transitions. Of course these would be more noticeable than four bit transitions, of which I am less confident.

The background hiss level seemed higher and less random in the 8 bit sections. The decay of tapped cymbals was harsher and truncated. Yada-yada-yada.


(results were here but I removed them so as to not influence others)

To describe things with impercise terms:

The main sounds of this music were not all that much changed when going from one bit depth to another. The sounds around the main sounds, however, were changed. Fewer bits, more blockier and harsh sounding with a less smooth decay. To me it sounds possible that with fewer bits, a lot of the low level detail is lost. Instead, there is false details created by the cruder 12 bit or 8 bit word depth.

Even is one is not able to consciously recognize a lower or higher bit depths, I supect listener fatigue will set in sooner with a lower bit depth.

I personally find that when I visit a high-end audio shop (or a low end), I usually start getting a headache in a few minutes in most cases. The times I don't is when I am listening to a good (good for me, anyway ... was it good for you?) system. I have gotten headaches from some very expensive and highly regarded equipment.

Paul, stilespj@mindspring.com

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the 1derful1

[This message has been edited by Paul J. Stiles (edited October 14, 2002).]

BoB-

1) As flat as I could get them by eq'ing out room induced peaks. Know how I knew there weren't any 6 dB rises due to the interaction at the crossover point? Because there weren't any major differences between sub alone, vs mains alone, vs sub+mains. Just the rises and falls due to the slopes on the filters applied to each. (Really neat to see in the real world, btw.)

2) I averaged the responses from where I sit, and the 2 other best listening positions. (Yes, I know what I'm doing.)

3) Very accidentally, I have a "good" room for acoustics. One side wall is open to the kitchen, so no reflections there. The other side wall is dominated by a sliding glass door, with a curtain in front of it. Carpeted floor. I even went so far as to put a make shift absorber in the corner of the room with the sidewall. One of those cardboard fabric bolts, covered by a few layers of fabric, propped in the bottom of the corner.

4) RS meter, every Hz from 20 to 98 Hz. With correction. Although to be honest, all the correction does is to change the tilt of the readings, doesn't hide any strange perturbations in the measurements.

Maybe instead of spending so much time trying to model these things with software, you should get out and do some real world measurements...

Kevin:

Just wondering. Do you have access to, or do you have an RTA program on your computer?

Well, I fully expected to post my results today, but unfortunately I could not get the cd to play. I have two dvd/cd player but both are Panasonic based (one is a Panasonic and the other is an EAD). I guess I could try it on my computer, but that I don't have very revealing speakers on my computer.


Let me know who to forward the CD to at rbessinger@deloitte.com

Actually there is a lot of discussion of this and related topics. Apparently it's not quite as cut and dried as that. Keep in mind the 'frequency response' of a speaker will say something like '1w/1m' but it's really a fixed voltage, not a fixed power, measurement. In designing crossovers for custom speakers there is quite a lot to take into consideration and the 'perfect' crossover seldom is really perfect.

I agree to the constant voltage in asmuch as it leads to a constant acoustic power which lead to a constant SPL. Electrical power will not be constant because of impedance variations of the speaker(s), however constant acoustic power is what we want inorder to get constant SPL. Of course, the desireablilty of high frequency beaming in order to reduce reflections to improve imaging and such will even modify the desirability of constant acoustic power.

Paul

Not all multichannel music is mixed this way although I wish it was. I think they use 6 full range speakers because I have at least four DVD-As that have high frequency stuff in the sub channel (Toy Matinee, America: Homecoming, Fleetwood Mac: Rumors, Eagles: Hotel California). They also mix stuff below 40 hz to the "5" speakers, especially when they place the bass drum in any of them. I believe the bass of a bass drum is below even 30 hz.

SH: Not yet. That is my next project, as soon as I get tired of doing all the manual measurements...

The mystery is revealed: the key to the timings of the changes in the "Listening Challenge" can be found at:

http://home.earthlink.net/~soundhound/timings.html

Did anybody hear all the changes? Be honest!!

[This message has been edited by soundhound (edited October 15, 2002).]

I was able to figure out that the changes were at 30 second intervals based on the beginning of the track. The 1:00 to 1:30 segment has a very audible increase in - now, no one get upset or worried - analog hiss. It is barely detectible from 0:30 to 0:31. So, I just guessed, rather than heard, that the changes were of the same duration. Once the tune starts moving, I found it difficult to hear any change. I was particularly listening to the ride cymbal and the two saxophones and thought I could hear subtle changes, but nothing definitive at all. As for Miles, well, I think you can record him at 4 bits or 2 bits and his tone will sound pure . I could hear no difference through the trumpet passage.

I was going to do some more listening tonight but I couldn't resist following the link. I don't profess to be an audiophile but I really appreciated this exercise. I'm going to tell my wife that I should have heard the difference as clearly as day and night and, since I couldn't, that I need to upgrade again. She'll tell me that it's a lost cause and try to talk me into the Lifestyle system from that company in Framingham, Mass.

Thanks Soundhound. BTW, how did your first subject do?

DollarBill:

Glad you had fun with it. I have to admit that 12 bit and even 8 bit doesn't sound as bad as would be imagined. The dead giveaway of course is the increased hiss of the 8 bit. That's the dither noise. BTW Gonk has a copy of the entire CD that I made with entire tracks quanitzed at the three bit levels and a big selection of originally 24 bit reverb tails, fades, etc at the three bit depths. It was intended to show how digital audio can sound when delibrately or accidently screwed up. I'm sure he could forward it to you.

Well, my friend was totally snowballed. But I have to admit I didn't tell him before hand that the track was doctored. I just played it all the way through for him, and then asked him if he found _anything_ amiss. He didn't. Then I told him he had been listening to 8 bit audio, and he didn't believe me, and still swore that he could instantly tell the difference between 24 bit and 16 bit stereo recordings of the same program. At the time I didn't have any 24 bit sources like I do now, so I just had to shrug my shoulders and suggest we go out for dinner, and of course a pitcher of Margaritas.

PS: Nah, the "I couldn't hear any changes" argument didn't work on my wife, but enough of the above mentioned Margaritas, and maybe some flowers.....well.....

[This message has been edited by soundhound (edited October 15, 2002).]

I'd agree with what Paul says. To keep myself honest I wrote down my impressions before going to the 'cheat sheet', and I wonder if maybe the original isn't as clean as it absolutely could be. Some places I could (still can) hear what sounds like old fashioned distortion, maybe the original master was oversaturated or something (or maybe my ears need looked at).

Anyway, the 8 bit sections have got notations in them like 'distorted','sounds like a kazoo','Bad again', etc.

Several places in the 16 bit sections I could hear someting not right, at least to me, but apparently that was either the way it's supposed to sound (I'm more a strings guy) or the original is not all it could be.

I was not able to clearly hear transitions, except for a few where we switched from 8->16 or 16->8, but there are notations in the middle if some sections where we went from 12->8 or 8->12 where I noted things.

I suspect an ABX of this and the original would be more productive, but it is a great demo, one that will stay in my collection.

Nice work! I'm sending it on to Matt Hill.

The original recording was done in 1959. The instrumentalists are Miles Davis on trumpet, Bill Evans on piano, Paul Chambers on Bass, John Coltrane and Cannonball Adderly on sax and Jimmy Cobb on drums. I'm not familiar with the details of the recording other than it being a live take.