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Audio Hardware / Re: Electrostatic speaker myths
Last post by Nichttaub -
Placement in the corner? My Staxes do not sound too impressive when I put them i the corner, no. (To those who didn't catch it: they are headphones ;-) )
So obviously I am not going to fill this post with the hard science or measurements the OP wanted (and by the way I think the Wikipedia article suffers from even more [citation needed]'s than is even indicated in the text).

Yeah, it reads like sort of vague advertising copy - very few of the assertions made in the article are backed up with facts.

Quote from: Porcus
One thing which comes up with maddening regularity is the so-called "massless" nature of the driver.  Obviously, this is pure hype because even the thinnest, most flimsy LDPE or Mylar films have a density of tens to hundreds of times that of normal atmospheric air.  Combined with the fact that the force behind the ESL "motor" is lower than that of a typical electrodynamic driver, some of the claims about the system seem questionable. 

Is it really so relevant to directly compare mass density to that of air? One thing is "density" vs "total mass", I mean, if you had the same mass but just made it thinner, you would inflate density, and ... that would not matter to the argument, ceteris paribus.
And it is not the heaviness of mass either - you wish to accelerate the driver [film|cone], so it would be inertia? But not directly compared to air; if one attempted to ask whether the inertia of cone / inertia of air relation is troublesome, and the inertia of film / inertia of air is small enough for that problem to be practically solved - then one would make the error of presuming that we want to move a lot of air as if it were a rigid body. (Gases are not - air waves are generating by attempting to compress it, right?)

Connected to that: a cone is "driven" where it is glued to the voice coil, and relies on sufficient rigidity for the surface. Electrostatic film is certainly not much rigid, and is driven everywhere.
But of course, this consideration is a matter magnitude that should be measured: if cone speakers manufacturers have solved the rigidity/inertia trade-off to an insignificant order of magnitude, then the return on removing the issue would not be too great.

Right; that's part of what I'm trying to determine.  You're moving a larger area but with (I suspect) much less force, and linearly over the vast majority of the surface but with a fraction of the excursion.  With a typical cone, the mass ratio is higher and the force much greater, but you have to have a greater excursion to achieve the same output level.  I was hoping to find some numbers to put that into perspective, so I could see whether the lower mass was a significant advantage or whether the ratios still came out similar.

I've heard some cone-type speakers which do a great job of reproducing transients and have the same sort of "hear-through" illusion that an electrostatic provides, so I have long felt it's not the transducer per se, but a combination of good drivers and crossover design (of course) plus other factors: room interaction, reflection/diffraction, and radiation pattern - all of which have common progenitors in the mechanical layout of a speaker but are usually not all well-behaved.

You can see in AJ's graph that the speaker represented has a lot of comb filtering at higher frequencies, and I'll bet there's a sort of "venetian-blind" effect from moving the listening position - plus probably timbral changes since the different frequencies don't change similarly at the different angles.  OTOH, they indicate a dipolar pattern which could be beneficial to eliminating room modes if positioned carefully.
Audio Hardware / Crown amplifiers for home audio?
Last post by JFS -
I am considering the crown amplifiers: XLS 1502 DriveCore 2 series.

Has anyone used these for home audio use? Would like to hear experiences. From what I have read on forums they are good and sound neutral.

A few things I have questions about:

1. Fans. Wondering if anyone has used these and if this is a problem, if so can it be fixed?

2. Preamplifier:
What would be good recommendations? The sensitivity can be set to 1.4Vrms or .775Vrms (this will lower the S/N by 6 db). Do I need to consider impedance when  matching the preamp and power amp? features? Hoping to find something economical: there is a used shop I can check out, and online.

3. Power output: How Crown measures watts
"The test is simple: the amplifier is set up with the level controls turned to the maximum setting, and then a connected sine wave source is increased until   the specified total harmonic distortion is reached"

I'm not sure how this compares to other amps. What would be the equivalent measuring 20-20khz?

I thought this would be a more economical way to get a amp than a HiFi shop. It can do well into 4 ohms.

Audio Hardware / Synching a 4th generation 64gb ipod
Last post by HA-User -

first, i did try my best to find a forum that i thought would best fit.

i had purchased a 32gb 4th generation ipod, and it was synching my requested playlists just fine.  but i bought a couple of 64gb ipods, figuring that i would eventually need them.

in itunes, i just finished creating one playlist that i wanted to sync.  but it was a couple gigs too big.  so i got out my 64gb, and tried to sync it.  it doesnt work.

i have approximately 10,000 mp3s loaded into itunes.  itunes finds them fine.  and my 32gb ipod will sync to what i ask it to do (until it got too large)

but when i hook up my 64gb ipod to itunes, it can only find 16 songs in the library.

so either it is looking at another library than my 32gb is doing, or there is something wrong.  both 64gb ipods do exactly the same thing, so i can rule out the ipod itself as the culprit.

for kicks, i did sync it to the 16 songs.  all 16 are part of my itunes library, but i see nothing to give me any clues.  these songs are not all in the same playlist, same artist, alphabetically close, etc.  it just seems as if it chose 16 random songs out of 10,000, and considers that to be my library !!

the ios is 6.1.6 on both the 32gb and 64gb ipods, and it says i have the latest version, if i try to update my version
3rd Party Plugins - (fb2k) / Re: JScript Panel script discussion/help
Last post by MordredKLB -
I would assume punctuation and other printable characters are the same, but the other things probably require a select statement based off the VK_ defines.
3rd Party Plugins - (fb2k) / Re: JScript Panel script discussion/help
Last post by davideleo -
on_key_down returns standard ascii character codes. So values 65-90 correspond to A-Z (you need to check if VK_SHIFT is down for upper/lower though). If you want to output the key pressed to the screen you can do something like:
Code: [Select]
var str = String.fromCharCode(65);    // str now contains 'A'

Yes, the A-Z character codes, as well as the blank space and 0-9 characters, are the same as the key code, but all other symbols are messed up and I believe they change with the keyboard settings.
Vinyl / Re: 33 vs 45rpm -- Technical Differences
Last post by Porcus -
But when analogue represents a continuous and interrupted signal irrespective of playback speed
Analog media isn't "continuous" as such, everything has a certain resolution, due to physical characteristics.

Analog-y in digital:,_performance_and_capacity
Why on earth spend 8" on 80 kilobytes of data when everyone above a certain age have had 1.44 megabytes fit on a much smaller area using the same principle? Because it is much more technically challenging to put the same signal on the same area (assuming you "read the surface", it is area).
Now fix the technology: area is proportional to the signal size you can write.
Audio playback is naturally constrained to 1 second of music per second (surprise!), so for a given tape or vinyl groove, lower speed means less area per second to carry the information, and less quality.
Support - (fb2k) / Re: Sample rate switching causes audio cutoffs...
Last post by arnaudf -
Thank for the answer, interesting :)
For Reaper : I set the Reaper's project sample rate (in Project Settings tab) according to the sample rate of the content. This way Reaper forces soundcard sr switch, if needed.

Indeed if a resampler dsp is used, there will be obviously no sr switch... but it's no more the originial content.
Will try with ASIO4all to see if it changes something.

I also tried with Winyl and observed the same behavior than Foobar (cutoffs).
Support - (fb2k) / Re: Sample rate switching causes audio cutoffs...
Last post by Case -
I didn't compile my own Audacity to get ASIO support so I tested REAPER instead. Granted I'm a total newbie with the program but I didn't see any option there to alter sample rate on the fly. The rate set in preferences was always used for playback and audio is resampled as required.

I tested my Asus soundcard's native ASIO and ASIO4ALL with the integrated sound chip on the motherboard (using Windows' driver). With REAPER I always lost the first samples on both sound devices and I tested WASAPI too and the same happened there. It also didn't matter if I used 44.1 kHz/48 kHz test file or 44.1 kHz/48 kHz samples. All combinations produced same results. And I did disable the fade effects from the file and the mixer showed action, just nothing came out of speakers.

With foobar2000 the native Asus ASIO drivers ate beginning of the samples when samplerate switched at track change. With ASIO4ALL beginning wasn't missing on either sound device.

But the very end of the previous track gets cut off with both ASIO implementations when samplerate changes. Same thing also happens with WASAPI event. WASAPI event over here allows hearing the very first samples of the next track though.
WASAPI push doesn't cut off the last samples on track change but it on the other hand snips away the beginning of the next track.

It seems like sound API/driver is doing the decision whether to play old buffered samples when stream format changes and just ignore new data until then or to drop buffered samples and start playing the new stream. It can't do both without something adding a delay.

Best solution to this problem is of course to use the DirectSound output method and utilize the existing resampler in Windows mixer. That way no samples will be missing in any situation. Another option is to use a resampler DSP so that DAC doesn't need to reinitialize itself in the middle of the playback.
Support - (fb2k) / Re: Long time present bug with pasting to playlist
Last post by Porcus -
Hold shift while pasting to forcibly paste at the end of playlist. This is hinted about in the statusbar when mouse hovering over 'paste' so it's not that hidden.
What about spending a menu item accessible with Shift + right mouse? Like, underneath Paste?

Thank you for your comment. Still I would be glad if simple "Ctrl+V" would paste new tracks always below selected item om playlist.
Then one would have a choice to make: above or below? I guess that most who select the first ("first of many", obviously) would want them pasted on top?
Support - (fb2k) / Re: [1.4 b3]
Last post by eagleray -
Then it appears the url doesn't return an actual audio stream or playlist. I'm guessing it's actually a web page with a player on it? What is the address?

It's a web page with a player on it.