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Bossobass Mini GTG Thread


Bossobass Dave

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How do you calculate reflection points?

 

But seriously, I took a mirror and walked it around the side and rear wall. Where I could see the reflection of the surround speakers is where I put the diffusion.

Using a mirror as described by Infrasonic here is probably a good starting approach.  The assumption is that the sound will reflect off of a large flat surface (e.g. a wall) just like light does with a mirror.  This is a fairly accurate assumption for high frequencies.  As frequency becomes lower, the sound exhibits more diffraction in which it bends and spreads out.  Low frequency reflections may involve a region of the surface rather than a single point.  For this reason, you'll have to cover a lot more surface area with bass absorption to completely eliminate a bass reflection because reflection from other parts will bend back to your listening position.  At the same time, you'll likely see better-than-expected results when covering a smaller area with bass absorption because a lot of bass actually diffracts into the absorber.  As such, precise placement of low frequency absorption is a bit less crucial.

 

You also should be on the look-out for secondary or tertiary reflections that may also arrive early enough to be relevant.  In my space, I had a problem with a ceiling reflection that also reflected off the back-wall before reaching the MLP, which was as strong as the reflection arriving directly from the ceiling.

 

My suggestion is to first obtain some broadband absorbers either from a company like GIK or by making them yourself from acoustic absorbent material like Owens Corning 703.  IMO, anything less than 2 inches thick is pointless for serious acoustic applications, and I recommend going at least 4 inches thick and using a small air gap between the absorber and wall surface.  Once you identify a candidate early reflection point, use REW to measure the room impulse response before and after temporarily installing an absorber centered at that point.  If you are comfortable doing geometry calculations, try to estimate the path length that the sound travels between the speaker and listener.  Take 1136 ft/s and divide it into the path length (in feet) that you calculated to estimate the arrival time of that reflection.  If you look at the impulse response or ETC in REW, you will hopefully see that a strong reflection was eliminated at the arrival time that you calculated.  If so, then you've found a reflection point that you'll probably want to treat.  If not, you may be dealing with multiple reflections that arrive at approximately the same time or maybe the point you trialed wasn't a strong reflection point after all.

 

Many experts advise treating early reflection points until the ETC no longer shows peaks greater -10 to -20 dB (relative to the initial peak at time zero) within the first 20 ms or so of the impulse response.  This is a very good approach to start with.  However, I find that the ETC and impulse response are too strongly biased toward the high frequency part of the reflections to be used exclusively.  Merely reducing ETC peaks to "-10 to -20 dB" probably isn't enough as it may leave the low frequency part of reflections untouched. 

 

I like to use REW's spectrogram feature to characterize the low frequency portion of reflections.  I typically use much shorter windows for these spectrograms than we typically use with SpecLab to show content on this forum.  The window size depends a lot on the frequencies of interest.  Lower frequencies require longer window times to get any usable data at all, but the window needs to be small enough window to distinguish the reflections from one another and from the direct sound.  If a reflection arrives in less time than it takes for a full wave at a particular frequency, then it may be hard to see even with the spectrogram.  For these, I often look at the frequency response while varying the IR window setting.  If the window is short enough to only analyze the response of the direct sound plus a single reflection, then you may be able to spot the damage done by observing the appearance of comb filtering as the reflection is included in the analysis window.

 

Anyway, there's a lot more to say than is reasonable for a single post.  If you want to discuss this further, we should create a separate thread for discussion.

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I'm not sure what you're saying here? Distortion /= Distortion of Original Shape? They seem like the same thing.    Up for interpretation? I don't think deviation from flat is up for interpretation. Any deviation is exactly that, a deviation. Taking an interpretation of frequency response is fine and dandy, but it's not an interpretation of the definition.

 

You say EQ is to fix things (yes I agree) and then challenge Bosso to try his subs without the shaping. So, EQ/shaping brings distorted (non-flat) systems into a format of less distortion (nearly flat) but Dave should try it without the shaping? That would keep the system in the distorted condition.

 

I think you're confusing terms here. The eq in Dave's system is to reduce distortion. Your reasoning is circular.

 

Umm, yeah. It was as confusing to me too so I probably miswrote a bunch of that. :P

 

Ok this is more coherent. The original dish is the intended, flat response. That's cool. If someone wants to hot sauce the dish, they're welcome to it, but it's not what the chef intended. You seemed to be saying the chef was putting grey matter on the table and people got to chose how the dish should taste. How it should taste and how they like it are different things.

 

 

Well now you get it. :D

 

Again, most of us start flat and then work our way from there. So it's not that I don't agree that flat is a good point to reach but sometimes if up to me as an end user to take my own creative license to it. I don't ignore that I am affecting the original response. I'm not that dense.

 

In my own experience, I go straight to getting as flat a response as I can. Then I listen. Usually the case is "there is not enough bass for my liking". So I turn up the bass. Now it sounds good. :)

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Using a mirror as described by Infrasonic here is probably a good starting approach.  The assumption is that the sound will reflect off of a large flat surface (e.g. a wall) just like light does with a mirror.  This is a fairly accurate assumption for high frequencies.  As frequency becomes lower, the sound exhibits more diffraction in which it bends and spreads out.  Low frequency reflections may involve a region of the surface rather than a single point.  For this reason, you'll have to cover a lot more surface area with bass absorption to completely eliminate a bass reflection because reflection from other parts will bend back to your listening position.  At the same time, you'll likely see better-than-expected results when covering a smaller area with bass absorption because a lot of bass actually diffracts into the absorber.  As such, precise placement of low frequency absorption is a bit less crucial.

 

 

I'm using a combination of diffusion and absorption in my room.

There is absorption behind my diffusors and a lot more absorption everywhere else.

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Yes. It's difficult to remember one-liners. Sorry.

 

 

I dislike any channel being 15dB above any other channel. It's gross distortion. Unmistakably audible. Far more so than and definitely distinguished from THD.

 

Was that the question?

 

We're on the same page.  Your treble and my bass are distorted because each have rolled off responses and are therefore not flat.  Got it  :)

 

Also, you're assuming that my CD/horn combo will be offensive to ones ears based on the frequency response I've posted.  I think you'd be pleasantly surprised.  

 

Your CC didn't bother me because of distortion, and I never suggested it was that.  I simply thought it was too loud.  As soon as I mentioned it you brought the LCR levels down a few db and bumped the bass up a few db, and it was fine from then on. 

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I've had a lot of experience listening to and playing along with musicians in informal settings, like a living room, garage, attic, porch, etc. It never once occurred to me or anyone else that the sound of instrument 'x' would be better if there were treatments on the walls, floor and ceiling.

 

Consider the instrument being played the recording. You may get up and sit in a different spot to listen. You may run around the room placing wedges, pillows, insulation, etc., to make the sound different. Change "the acoustics" of the room to some infinite point between an empty church and an anechoic chamber.

 

But, you aren't changing the "recording". You can't change the SQ of the instrument by "treating" the room.

 

Since sound recording and reproduction began, there have been people studying and teaching the ideal aoustics for the listening environment. It makes sense to me to avoid "double reverberation", but it's similar to post smoothing EQ, IMO. Season to taste. I've always shied away from those discussions.

 

The discussion on low end reproduction of modern recorded source through a summed mono signal of redirected bass from 'x' discrete channels and a discrete LFE or .1 channel has been ongoing for 15 years. That's enough for me. :P

I'm not sure what you are getting at here.  No, room acoustics do not change the sound produced by the instrument.  Room acoustics definitely do impact what we hear or what a microphone records when an instrument is producing sound in that room.  When doing actual recording, close micing can help reduce the contribution of the room to the sound, but this technique isn't fool-proof and has its own set of problems.

 

And yes, there's plenty of subjectivity as far as what kind of room acoustics are most desirable for different kinds of sound, but there's plenty of room for objectivity as well.  Thinking about speakers, we all may have different preferences for what kind of speakers sound best, but we can probably all agree that certain speakers sound much better than certain other speakers.  Likewise with room acoustics, many changes are possible that almost all listeners would agree are positive or negative.

 

There's also a lot more to acoustics than just decay time as you allude to with your statement about changing acoustics to "some infinite point between an empty church and an anechoic chamber".  Decay time typically varies with frequency, and the quality of the reflections matters a lot.  Moreover, variation in room acoustics and how the speakers interact with the room likely has a lot to do with what kind of non-flat target curve people prefer for post-smoothing EQ.

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In my own experience, I go straight to getting as flat a response as I can. Then I listen. Usually the case is "there is not enough bass for my liking". So I turn up the bass. Now it sounds good. :)

 

Often, the pros do exactly this sort of thing too.  When setting up a new system or a new room, they start flat, listen to some reference material, and then adjust the target curve until it sounds right.  The fact of the matter is that the notion that a flat response and a certain playback level (e.g, calibrated at 85 dBC or rather 83 dBC as it probably should be) iare correct and everything else is distortion is a myth.  Reality is much more complicated, and different speaker / room acoustic combinations will yield very different subjective results with the same playback level and target curve.  This is the translation problem which is fairly well known among audio professionals.  It is furthermore observed that changes made to both playback level and target response curve can help achieve a closer match to what was heard in the original mixing environment.

 

With that said, differences between treble and bass of 15 dB seem a bit extreme to me.  Certainly, if male voices sound muddy, then you might have the bass turned up too high.  OTOH, adding some bass absorption may help a lot here too.

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Often, the pros do exactly this sort of thing too.  When setting up a new system or a new room, they start flat, listen to some reference material, and then adjust the target curve until it sounds right.  The fact of the matter is that the notion that a flat response and a certain playback level (e.g, calibrated at 85 dBC or rather 83 dBC as it probably should be) iare correct and everything else is distortion is a myth.  Reality is much more complicated, and different speaker / room acoustic combinations will yield very different subjective results with the same playback level and target curve.  This is the translation problem which is fairly well known among audio professionals.  It is furthermore observed that changes made to both playback level and target response curve can help achieve a closer match to what was heard in the original mixing environment.

 

Well said, SME. It's definitely more complicated than often mentioned.

 

With that said, differences between treble and bass of 15 dB seem a bit extreme to me.  Certainly, if male voices sound muddy, then you might have the bass turned up too high.  OTOH, adding some bass absorption may help a lot here too.

 

Fine.... I'll go redo my speakers so they're back in boringsville. :P

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Tux gets it. It's a simple matter of a simple interpretation of a simple term that everyone at the level of this forum should easily get without taking it personally.

 

Right, I don't worry about little babies toes when I post because I assume no little babies are reading my posts. It's a posting character flaw. One I'm not gonna do anything about. Sorry, this is the best I can do.

 

But, being completely honest about it, people who jack a thread with

 

 

Your strong opinion that sealed is the only palatable answer and frequency responses below 15hz is vital, is simply that --- your opinion.  I personally add my name to the list of those that disagree here.

 

Dude, WTF? I said nothing to the guy, nothing about the guy, about his listening preferences nor any mention of him at all. It's like, "You ain't all that because I don't like any content under 15 Hz and all my friends agree with me, not you. That's right, 15 Hz. Not 14 Hz, not 16 Hz, not even 14.9 Hz..." You don't see how absolutely inane that bullshit is? I don't recall ever jacking a ported subwoofer thread to piss on the OP. Start a thread... oh, wait, he has started a plethora of threads and mentioned the same audio bias in all of them.

 

Now, I'm not about to go running to the principal with tears in my eyes, but if you expect me to answer a post that starts out with that sort of sentence all PC and shit, you're in Wonderland riding a Unicorn. B)

 

Yes, I get the palate thing, but... I don't discuss palates. ;) Hot sauce, whipped cream, who cares? I just dislike when the subject jacks my thread. It's as irrelevant to any serious discussion about audio as it can be.

 

And, Luke, what am I gonna do with the stuff you're layin' on me? A 6dB tilt across 19 thousand nine hundred Hz being equated with a 4,000dB/octave shelf with 20dB of boost across eighty five Hz?

 

Why do people use post PEQ to pull down peaks? Because the peaks are audible distortions that sound bad. But, if it's a 20dB bandpass hump created by a HPF, shelf boost and LPF, it sounds good? No it doesn't. It sounds loud. Deafeningly loud. There are no other redeeming qualities except for you to say you like it that way. Good enough. I thought we more than covered that in another thread? :mellow:

 

SME, exactly what I'm saying is that room treatments are in the same basket as preferred listening levels, CD driven horns vs dome tweeters, dome tweeter arrays, electrostatics, ribbons and subwoofer bandwidth and level calibration preferences. Hot sauce. B)

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BTW, I'll spare anyone wondering the suspense...

 

Dave, you're a guy I could sit and talk to all day about audio in person, but your personality on message boards makes it hard to even read your posts anymore.  There, I said it.  

 

 

Now that that's out of the way, here's what bothers me.

 

1.  Your way to rationalize the importance of single digit response is that it's "simply accurate vs other".  In other words, flat bass response to the single digits equals accuracy and anything else doesn't.  Here's a quote from one of my threads:

 

"And, please everyone... no one is telling anyone else what to like or dislike. If the discussion goes there, the discussion becomes a din of nonsense. And, over the years, this is the straw man used... accusing me of dictating what people should like. I could not care less, honestly and truly, what anyone else, everyone else, the majority of the world's population, etc., likes or dislikes.

 

It's simply accuracy vs other."

 

If accuracy is really your driving factor, then why does this only apply to bass and not treble?  If you really care about accurate response, why ignore the roll-off you have on the top end?

 

While at your house we had a good conversation about my experiences on a concrete floor and very capable sealed systems, with nothing below 15hz being heard or even felt.  The only indication of single digit activity what was visually seeing the cones move.  Although I'm sure you were/are skeptical, you didn't discredit my experience.  

 

Asking the same question on Adam's thread, here's your response (http://data-bass.ipbhost.com/index.php?/topic/284-bossobass-raptor-system-3/?p=4725):

 

"My answer is that it is virtually an impossibility that it will do nothing to my senses."

 

So my experiences were somehow an impossible scenario, or I'm lying, or what?  

 

So yeah, you're a different person on the boards vs in person, and this has nothing do to with "accuracy", but rather your home theater experience in your room with a 7hz resonating floor.  If you take your same subs with the same frequency response and same SPL levels in a room with a concrete floor the experience WILL be different.  

 

People like what they like, and I'm obviously no exception, but I'm not going to justify my subjective preference in the name of "accuracy".  I don't buy that angle for second.    

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Fine.... I'll go redo my speakers so they're back in boringsville. :P

As you said, you like hot sauce.  There's nothing wrong with that.  I haven't heard your system, but I'm guessing I'd like it more if the bass boost was more gradual between 50 and 200 Hz.  I like upper bass too, and the 80-200 Hz range contributes a lot of tactile punch and sense of speed to the experience.  Then again, 100 Hz at +10 dB would likely cause the vocals to get too chesty.  Enjoy your hot sauce.  :)

 

SME, exactly what I'm saying is that room treatments are in the same basket as preferred listening levels, CD driven horns vs dome tweeters, dome tweeter arrays, electrostatics, ribbons and subwoofer bandwidth and level calibration preferences. Hot sauce. B)

 

I understand what you're saying.  I totally disagree.  Look at any unsmoothed frequency response measurement and tell me how accurate it looks.  Unlike typical electronics and even the speakers (if they are decent), the room grossly distorts the time and frequency response of the sound.  Room treatments substantially reduce that distortion thus improving accuracy.

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Contrary to Dave's belief that we must all adhere to 'flat equals signal reproduced faithfully' most people do not like the sound of 'flat'.

I do. The only time I don't prefer flat is when I listen to a crappy mix. Most studios aim for flat because it gives you a consistent playback from system to system. So if your intent is to listen to the artist's song the way that they intended you to hear it, your aim should be for flat reproduction. If you like to remix the stuff with your system, that's cool too but it's not like 1 curve is gonna get it all. You'll have to tweak each production house's stuff until it sounds close to the way you like it. I used to do that but it turns into a full time job fast. Now if the mix isn't up to par I push the stop button and find something else that I like. Tons of good mixes out there, just have to sift through some junk to find it.

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Getting back on track here...As promised, here's a review of what we've dubbed the Raptor-SI. The SI, of course, is an acronym for Stereo Integrity and is attached to the Raptor name because this particular Raptor is loaded with two SI model HST-15 drivers.

Nick, the proprietor of Stereo Integrity, whipped up the pair on short notice and personally lugged them down to Charlotte for the GTG. He left them here so that we might run a few tests and let him know the results. He not only doesn't mind that the results are posted publicly, he prefers it, being fully confident in his "babies" capabilities. Based on their performance with a few soundtrack clips, I see where he's coming from. :)

As most members and guests here already know, we test subwoofers using snippets of actual soundtrack as input signal. Recently, Max sent Paul his first draft WCS (Worst Case Scenario) test signals. The disc consists of numerous sine wave tones from the subwoofer bandwidth, mixed at nearly 0dBFS in 7.1 channels which we then redirect to the subwoofer channel and use some of those various tones as input signal for additional testing.

We also have received the first of several new amplifiers intended to power single 2 x 15" Blackbird and Raptor modules, so we decided to use it to power the Raptor-SI for these tests.

So, here's the scoop:

Device Under Test (DUT):

A single Raptor module loaded with two Stereo Integrity HST-15 drivers in a dual opposed configuration. This model is equipped with the custom machined aluminum grill ring, stainless steel screw covers, hand polished stainless steel top plate hardware and 1/2" thick, beveled glass top plate. It has stainless steel adjustable legs and stands 33" tall with an overall diameter of 22-1/2". The Raptor-SI weighs in at approximately 180 pounds.

 

tf0OwZ9.pnghttp://i.imgur.com/tf0OwZ9.png



The amplifier is Class D and the specs are shown in the pic below. It is available in a regular model and a model with on board DSP. We haven't yet tested the DSP section, so this review only contains data on the amplifier's performance as tested.

 

The Frequency Response Transform process:

 

This is to show how the signal shaper transforms the native frequency response of the subwoofer into a new frequency response with the desired F3 and Qtc so that it will best mate with the room's boundary gain transfer function. Many people have confused this design step as being the same as applying post smoothing equalization to affect the response at the seats. The transform from native to designed frequency response of the sub to mate with the boundary gain transfer function is wholly unrelated to EQ tweaks made to tame room mode peaks and other frequency response non linearity at the listening position. This review doesn't allow time to go into the differences in detail.

 

Of course, we could not have had a specific SEQSS model designed and built for the GTG for the Raptor-SI in advance, so, instead, we used the Marchand Bassis to dial in a specifically shaped signal based on a) the native response of the Raptor-SI and B) the average room gain profile curve.

 

The goal here is, as stated above, to arrive at a new native frequency response from the subwoofer that has a response that mates with the room gain curve, including the proper Qtc. In addition, it must agree with the available power, power handling and required power for the specific sub.

 

QZEjJ0h.png

 

^^^Mission Accomplished^^^

The Player => Processor => Amplifier Signal Chain and Gain Stage:

We use the OPPO BDP-105 as a player/processor. The OPPO decodes the digits, processes them, converts them to analog and sends them to the various cannel's separate amplifiers in the system through it's analog outputs.

 

Marchand Bassis loop/Entire Signal Chain Loop animation:

 

9d8996a1dbab689d434f6e13286950d2.gif

 

The final signal chain to the Raptor-SI module is shown in green and is analyzed in the graph above the animation.

As some may know, Paul recently found through tests on the BDP-105 that it cannot properly process the summing of the LFE channel with redirected bass from any channel without going into severe clipping. His tests showed that the OPPO can process the LFE channel with no clipping, but when any channel of redirected bass is summed with the LFE channel, clipping occurs when the subwoofer trim is set anywhere above '0'. It does not matter what the Master Volume is set to and the clipping is severe in cases where the soundtrack is mixed hot. Although the soundtrack clips we used were a Worst Case Scenario, unfortunately, every soundtrack we use for tests can be categorized as being mixed hot enough to cause the problem in the OPPO.

Paul has measured the voltage out of the player ==> SEQSS using all of the bass-heavy scenes in our movie library. This constitutes the actual real Worst Case Scenario requirement in our library of movies, so these scenes are the ones he uses to gauge the reaction of the DUT under normal use.

Of late, there has been much ado about members who like to run their subwoofers 'hot'. This is normally accomplished by bumping the subwoofer trim after calibration. Since reference calibration is a fairly loose term, the bottom line is that few really know how much more elevated the playback level of their subwoofer is over the satellites. But, the consensus is 'by a lot'. If, for example, the subwoofer is run at calibration plus 15dB, it means that the mixed subwoofer level is bumped by some 32 times original intent.

Of course, for these tests, a metric of some sort must be the goal. The reference we use is what is referred to as "Reference Level". The OPPO is set to reference Master Volume Level with the sub woofer trim set to '0'. The SEQSS has been designed to translate that level with the best signal to noise possible (>116dBA). The amplifier's gain is then matched for the best S/N in the chain using it's gain range settings and front panel gain attenuator. The resulting level is matched using the amplifier's gain attenuator controls. This tells us if a subwoofer can handle the truth. Reference level playback with the system calibrated flat with only hard limiting in place. Beyond that, whether or not the sub can withstand "running hot" is impossible to say as it's an arbitrary term that can only be determined by the end user.

Below are the resulting frequency response magnitude graphs of a) response after signal shaping,  B) response at the seats before tweaks and c) d)  e) response at the seats after digital delay and volume leveling. No post smoothing EQ was used in the tests. The final response at the seats was approximately 3-200, (+/-) 5.5dB.

 

1C903ml.png


 

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Reserved for Paul's test results:

 

I'll post the new amplifier data here and Paul can add the data he collected during the tests.

 

Here's a pic of the A5K and A5K DSP with the frequency response and specs. The DSP stuff is controlled via a USB jack to a PC. This amp has not been sold in the US before and the drivers for the software have not yet been provided us by the manufacturer, so I can't show the GUI or any specifics of the DSP yet. I'll add that info as it becomes available in the next week or so.

 

It's solidly built, nicely laid out, fairly quiet fans, has a 1/4" solid machined aluminum FP and works as advertised, so far.

 

xxNWQUa.png

 

Here also is the results of the Raptor-SI system during reference level playback of a difficult scene from the recently released Jupiter Ascending:

 

do1FyrV.png

 

4ac45b1ba095da31635dee804a78a105.gif

 

The sub stayed intact through repeated playback of this scene. I believe the drivers were at maximum usable throw and response held up to 5 Hz. I believe that a corner placement will probably increase extension to just <4 Hz based on previous experience and the data collected on this sub.

 

As it was, the sub compressed during the hardest hit. The FR is down -6dB or so at 20 Hz, but there is still compression where the digits show light blue and the mic'd sub shows gray. I also heard a slight mechanical distress at the transient with my ears clamped shut, close to the top driver. I would estimate that this 2 x 15" sub will handle the hit at -5dBRL in my 3500 cubes room, corner loaded. When more folks give this movie a spin, I imagine there will be more appreciation for the performance of this sub. It's a truly brutal hit at 25.5 Hz but with a wide spread of freqs, all the way down to below 3 Hz. You know, the stuff we live for and love to discuss.

 

Paul will add the volts/watts/amps data that the A5K DSP threw at the Raptor-SI when he gets the time. Stay tuned...

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Reserved for Tim's pics, which really give a good idea of the aesthetics of the Raptor sporting HST-15s...

 

Man, it's awesome to see someone take the very same tool you tried to use many times and use it properly to get to the bottom line. Tim also decided to have some fun while snapping pics of the Raptor-SI with both the MDF top plate option and the glass top option:

 

whImOac.png

 

This one's called the Gambler:

 

YnPTIur.png

 

This one's called WAF:

 

0Ivl2BM.png

 

Full-on shot of the Raptor-SI with MDF top plate and close-up:

 

FK913H7.png

 

Thanks, Tim!! Awesome job. B)
 

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I do. The only time I don't prefer flat is when I listen to a crappy mix. Most studios aim for flat because it gives you a consistent playback from system to system. So if your intent is to listen to the artist's song the way that they intended you to hear it, your aim should be for flat reproduction. If you like to remix the stuff with your system, that's cool too but it's not like 1 curve is gonna get it all. You'll have to tweak each production house's stuff until it sounds close to the way you like it. I used to do that but it turns into a full time job fast. Now if the mix isn't up to par I push the stop button and find something else that I like. Tons of good mixes out there, just have to sift through some junk to find it.

 

The idea is to have everyone calibrate flat to get consistent playback.  The trouble is, it doesn't work.  The same smoothed frequency response can sound very different in different listening environments.  Exactly how systems should be calibrated for consistent playback is an open problem, but there's a fair amount of psychoacoustic knowledge to justify the fact that flat is not always right.  This HTS thread with contributions from Bob Katz discusses a fair bit of theory.  I believe he uses a 6 dB log-linear roll-off from 1 kHz to 20 kHz.  Another on that thread has a 10 dB log-linear roll-off from 20 Hz to 20 kHz.

 

Yes, it's a big pain to tweak your target curve for every bit of content, but it's certainly worth tweaking it once for good results with the widest range of content.   As Bob points out, often movies benefit from another 1-2 dB treble roll-off on his system.  That makes sense because the acoustics at dub stages are quite different from the acoustics in typical music mastering rooms.  This isn't necessary for everyone though.  Now that I have my room treatments including diffusion and my curve is flat from 250 Hz and up, I don't notice significant treble difference between music and movies.  I wonder if Bob were to add diffusion (AFAIK, he only has absorption) if he'd end up with a curve with only bass boost, more like myself and infrasonic?

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Dave,

 

Thanks for posting all the data!  I have some questions.  Is the A-7k really 4 channel?  From the specs you posted, it looks like the A-7k can do 14kW total just like the A-14k.

 

Now about that DSP, can you find out what its internal precision is?  Lots of audio DSPs (such as MiniDSP 2x4) are 56-bit fixed point, and this works well enough for all but the lowest frequencies.  At least with the MiniDSP, the DSP precision becomes a problem once you start trying to do anything below 30 Hz or so, and it is practically unusable below 20 Hz.  A 32-bit floating point DSP is more expensive but offers a lot more usable precision at very low frequencies.  Also, does the DSP allow custom biquad input, like the MiniDSP products do?  This feature along with sufficient internal precision could allow all the signal shaping to be done in the DSP, which is a real nice option.  Lastly, will the DSP feature be available with the A-7k?

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Now about that DSP, can you find out what its internal precision is?  Lots of audio DSPs (such as MiniDSP 2x4) are 56-bit fixed point, and this works well enough for all but the lowest frequencies.  At least with the MiniDSP, the DSP precision becomes a problem once you start trying to do anything below 30 Hz or so, and it is practically unusable below 20 Hz.  A 32-bit floating point DSP is more expensive but offers a lot more usable precision at very low frequencies.  Also, does the DSP allow custom biquad input, like the MiniDSP products do?  This feature along with sufficient internal precision could allow all the signal shaping to be done in the DSP, which is a real nice option.  Lastly, will the DSP feature be available with the A-7k?

It says "24 bit/96 kHz System Processor" on the front of the amp which I take to mean it does all processing at 24 bits. 

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It says "24 bit/96 kHz System Processor" on the front of the amp which I take to mean it does all processing at 24 bits. 

Most likely that means that the ADC and DAC sample at 24-bits 96 kHz.  The internal precision will be higher in any competent audio DSP design.

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Dave,

 

Thanks for posting all the data!  I have some questions.  Is the A-7k really 4 channel?  From the specs you posted, it looks like the A-7k can do 14kW total just like the A-14k.

 

Now about that DSP, can you find out what its internal precision is?  Lots of audio DSPs (such as MiniDSP 2x4) are 56-bit fixed point, and this works well enough for all but the lowest frequencies.  At least with the MiniDSP, the DSP precision becomes a problem once you start trying to do anything below 30 Hz or so, and it is practically unusable below 20 Hz.  A 32-bit floating point DSP is more expensive but offers a lot more usable precision at very low frequencies.  Also, does the DSP allow custom biquad input, like the MiniDSP products do?  This feature along with sufficient internal precision could allow all the signal shaping to be done in the DSP, which is a real nice option.  Lastly, will the DSP feature be available with the A-7k?

 

Nice catch, thanks! I fixed it. Both amps are 2 channel only.

 

No sepcifics on the DSP yet. Updates as they become available and if we decide to offer them. :)

 

EDIT: I see Shred beat me to it on the DSP Qs.

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Wow, didn't anticipate missing a few days and seeing all THIS goodness going down. I agree with what everyone has said in the last 4 pages basically. Like no really, I agree with basically everyone and think that a lot of it is simply semantics. That's as far as I'm going with it :D I love you all!  ^_^

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As you said, you like hot sauce.  There's nothing wrong with that.  I haven't heard your system, but I'm guessing I'd like it more if the bass boost was more gradual between 50 and 200 Hz.  I like upper bass too, and the 80-200 Hz range contributes a lot of tactile punch and sense of speed to the experience.  Then again, 100 Hz at +10 dB would likely cause the vocals to get too chesty.  Enjoy your hot sauce.  :)

 

 

I understand what you're saying.  I totally disagree.  Look at any unsmoothed frequency response measurement and tell me how accurate it looks.  Unlike typical electronics and even the speakers (if they are decent), the room grossly distorts the time and frequency response of the sound.  Room treatments substantially reduce that distortion thus improving accuracy.

 

 

Absolutely. And I'll take it one further, it starts at the wall studs. A strong quiet skeleton goes a long, long way towards improving all sound top to bottom, in a huge way. 

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Reserved for Tim's pics, which really give a good idea of the aesthetics of the Raptor sporting HST-15s...

 

Man, it's awesome to see someone take the very same tool you tried to use many times and use it properly to get to the bottom line. Tim also decided to have some fun while snapping pics of the Raptor-SI with both the MDF top plate option and the glass top option:

 

whImOac.png

 

This one's called the Gambler:

 

YnPTIur.png

 

This one's called WAF:

 

0Ivl2BM.png

 

Full-on shot of the Raptor-SI with MDF top plate and close-up:

 

FK913H7.png

 

Thanks, Tim!! Awesome job. B)

 

Damn, Tim's pretty good at taking pictures now!

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