I tend to agree. I've used a PC for most of my music listening with plain olde WMP and the rather crappy onboard sound card and the results are great! The CD player has less noise but other than that, if I use tracks converted as 'lossless' it's hard to tell the difference. It's so great to customize music to your taste. For instance, Men at Work's 'It's a Mistake' is super-bright in the mid to upper treble. A few mouse clicks and I can roll it off and voila. Or take Staind's 'Epiphany'. It's got way too much low bass thump for me. I can roll that off in seconds as well.
Oh ha, just the little cheesy one in WMP. I can't boost bass, though. It leads to incredible distortion, even at mild levels. Actually, I don't even know if what I'm doing is a good idea. I have no idea if there are any impedance mismatches or other issues. Gear: Via AC'97 sound card (old). Output is stereo of course, I split it to two mono audiophile rca interconnects. These are run into my old Denon PMA S1 integrated amplifier.
I've never bothered to ask anyone 'in the know' if this is a good idea, but it has sounded good for years. The only issues are the inability to boost bass and a healthy amount of high frequency hiss coming from the card/pc.
I've never bothered to ask anyone 'in the know' if this is a good idea, but it has sounded good for years. The only issues are the inability to boost bass and a healthy amount of high frequency hiss coming from the card/pc.
AC'97 implementations tended to be really really bad.
Almost anything available today would be much better. I prefer the USB sound options, because it gets the analog stuff outside of the rather noisy environment of the PC case.
There are many many USB sound cards, pick any one and it should be a big step up from AC'97.
Almost anything available today would be much better. I prefer the USB sound options, because it gets the analog stuff outside of the rather noisy environment of the PC case.
There are many many USB sound cards, pick any one and it should be a big step up from AC'97.
as said an usb DAC can and should be used instead of the soundcard, i was looking at the music streamers... is this the source that drives your openbaffles??
i think both WMP and iTunes eq should be avoided. so does the core-audio eq used under osx trought a wst wrapper for i-Tunes (altought better than the built-in)
WST plugins are much better! many of them use 64bit precision and lot of configurable filters. the one i use electricQ is really trasparent, and the shareware version also has linear-phase. believe me you will hear the difference over WMP and Itunes built in.
also i can tell the difference when i swap from normal playback to WASAPI mode (windows audio session api, while OSX and linux always had a good audio core for musicians use them in live playback) that's on my macbook built-in audio output
just download foobar, the WST wrapper component and the WASAPI component, and copy them in the component directory... then preferences and turn them on... if you need help write me.
also don't turn the volume down from the computer.
i think both WMP and iTunes eq should be avoided. so does the core-audio eq used under osx trought a wst wrapper for i-Tunes (altought better than the built-in)
WST plugins are much better! many of them use 64bit precision and lot of configurable filters. the one i use electricQ is really trasparent, and the shareware version also has linear-phase. believe me you will hear the difference over WMP and Itunes built in.
also i can tell the difference when i swap from normal playback to WASAPI mode (windows audio session api, while OSX and linux always had a good audio core for musicians use them in live playback) that's on my macbook built-in audio output
just download foobar, the WST wrapper component and the WASAPI component, and copy them in the component directory... then preferences and turn them on... if you need help write me.
also don't turn the volume down from the computer.
No, this lame PC setup is certainly NOT for critical listening as a modded CD player stands by for that job. And yes, it's in dire need of upgrade. I just subscribed to the notification list for the assembled Monica when it is back in stock.
The PC setup is used for my 600 or so lower-quality MP3s that I stole using filesharing years ago. Some of them have surprising SQ nonetheless. It's also used for games and regular PC sound. And yes, the OBs are used for everything. The fullrange drivers supposedly need hundreds of hours of 'break-in' so I give them everything I can.
And thumbs up to you guys for offering helpful advice instead of jeers (and this setup of mine deserves jeers!)
The PC setup is used for my 600 or so lower-quality MP3s that I stole using filesharing years ago. Some of them have surprising SQ nonetheless. It's also used for games and regular PC sound. And yes, the OBs are used for everything. The fullrange drivers supposedly need hundreds of hours of 'break-in' so I give them everything I can.
And thumbs up to you guys for offering helpful advice instead of jeers (and this setup of mine deserves jeers!)
i think audio from the pc is just as serious as one wants to take it. and in the coming years it will definitely win over cds. there is already some recording company that is offering downloads of master quality stuff (at very high sample rates), it will take just some time to free us from physical supports and their limitations. only vynil will still be the contender on SQ.
how is it gaming trough the OBs? do the woofers handle well the LF content of games?
how is it gaming trough the OBs? do the woofers handle well the LF content of games?
Could you explain why?
Well my OBs are not purist. A large bass guitar speaker handles the upper bass and a large PA speaker handles 30-80 on its own amp so I can have as much bass as I want for games or movies. For music, I often leave the 'sub' (I hate calling it that) off as the bass guitar speaker does so well on its own down to 60hz with room gain.
the volume control on the pc is said to work in a digital fashion so that the lower the volume, the lesser the bits, the lesser the dynamic, it does not work in a trasparent way, it just cuts the value to the nearest one... you don't want it after a 64bit precise equalizer... anyway i don't know enough about it, and i often do it, but not when i'm on serious listening... than it's a been a long time that i'm not on serious listening since i broke my audionirvana (the ones a year ago i asked you suggestion about OB) and i still have to have my own alpair bvr done.
Ok I'll take your word on that.
How did you break the ANs? I torture mine daily and they seem to be tough as nails. Unfortunately, they also will sound like nails if I use them outside their intended application (loud rock music) but I do it anyway.
How did you break the ANs? I torture mine daily and they seem to be tough as nails. Unfortunately, they also will sound like nails if I use them outside their intended application (loud rock music) but I do it anyway.
from the internet about k-mixer (the winXP audio core), i don't know if it holds true for win7:
...the mixer of windows resampling process worsens the audio quality. The windows plugins like Volume, Balance etc.. are only working with 48kHz music signals. Here an example on how windows processes the audiostream: input signal CD 44.1kHz, upsampled to 48kHz for the windows plugins, then downsampled to the origin 44.1kHz. Even if plugins like volume are set to maximum level, the resampling process takes place.
Even if the output is bit exact, the resampling algorithms may induce jitter resp. signal alterations.Better: when listening without ASIO, turn the volume in K-Mixer to maximum and adjust volume in your hifi set!
Volumechanges done in Computers or HiFi devices, based on changes of the digital Audiosignals, resulting always in loss of dynamics.The K-Mixer resampling algorithm has a negative impact on the sound quality across the whole acoustic bandwidth and dynamic spectrum:
* Reduced room information (transients, ambience, depth)
* Loss in fine dynamics
* Loss in precision
* Inferior definition and depth of bass
* Reduced transparence
An ASIO driver bypasses the K-Mixer, this "audiophile evil". When audio output is routed through the ASIO driver, Windows can no longer manipulate it. For example, you can no longer change the volume in the audio player software or in Windows.Note that not all listeners find a striking difference between audio output via ASIO and the K-Mixer. Other important factors in perceiving audible differences are:
* The quality of the audio source
* Other equipment in your Hi-Fi setup, e.g., amplifier/speakers
* Individual ear training and hearing capabilities
* Individual audio "taste" and listening preferences
anyway ASIO is not usefull anymore for playback now that there is WASAPI (that seems a japanese food)
...the mixer of windows resampling process worsens the audio quality. The windows plugins like Volume, Balance etc.. are only working with 48kHz music signals. Here an example on how windows processes the audiostream: input signal CD 44.1kHz, upsampled to 48kHz for the windows plugins, then downsampled to the origin 44.1kHz. Even if plugins like volume are set to maximum level, the resampling process takes place.
Even if the output is bit exact, the resampling algorithms may induce jitter resp. signal alterations.Better: when listening without ASIO, turn the volume in K-Mixer to maximum and adjust volume in your hifi set!
Volumechanges done in Computers or HiFi devices, based on changes of the digital Audiosignals, resulting always in loss of dynamics.The K-Mixer resampling algorithm has a negative impact on the sound quality across the whole acoustic bandwidth and dynamic spectrum:
* Reduced room information (transients, ambience, depth)
* Loss in fine dynamics
* Loss in precision
* Inferior definition and depth of bass
* Reduced transparence
An ASIO driver bypasses the K-Mixer, this "audiophile evil". When audio output is routed through the ASIO driver, Windows can no longer manipulate it. For example, you can no longer change the volume in the audio player software or in Windows.Note that not all listeners find a striking difference between audio output via ASIO and the K-Mixer. Other important factors in perceiving audible differences are:
* The quality of the audio source
* Other equipment in your Hi-Fi setup, e.g., amplifier/speakers
* Individual ear training and hearing capabilities
* Individual audio "taste" and listening preferences
anyway ASIO is not usefull anymore for playback now that there is WASAPI (that seems a japanese food)
Last edited:
i was having some fun trying them alone on large baffles, while choosing which woofer to buy, the window was open and the wind pushed one baffle face to floor, the frame of the driver collapsed under the magnet of about 1/2 cm... they are big magnets, aren't them?
btw foobar is just a usefull and clean player. it does not sound better than any other player, but it's meant to be expandable: it supports components, and among them there is this wonderful possibility of wrap the signal to professional VST DSP like the equalizer i'm using, and does it sounds better!
btw foobar is just a usefull and clean player. it does not sound better than any other player, but it's meant to be expandable: it supports components, and among them there is this wonderful possibility of wrap the signal to professional VST DSP like the equalizer i'm using, and does it sounds better!
Oh bummer. Yea the AN 8 uses a 51 ounce ceramic mag. A cast frame would have cracked; the stamped steel one will bend. Either way, it won't be easy to sell one AN driver.😀
I will try the fubar software also.
I will try the fubar software also.
i must say David at CSA was kind enough to offer to send just one driver, but with the expedition costs it just didn't take sense... rightnow makes for a funny mono system with a baffle of about 110cm, clipping at any serious bass non the less... and heavy equalization...
... and yes foobar has a downmix to mono dsp also, altought it someway muddle the sound
if you give it a go remember to check out electriQ
... and yes foobar has a downmix to mono dsp also, altought it someway muddle the sound
if you give it a go remember to check out electriQ
This thread is interesting. I'm attracted by the idea of equalising some small sealed speakers flat. Whilst I'm sitting at my PC, I might as well use it's processing power.
FOOBAR offers the easy route I suppose. But there are also EQ plugins available for the squeezebox which would be cool so I can stick with softsqueeze (the software version of squeezebox).
The flat to 30Hz Alpair5 was interesting LEANAUDIO: Report of Alpair5 Aplication but is it serious? What SPL do you need for quiet working at your PC?
FOOBAR offers the easy route I suppose. But there are also EQ plugins available for the squeezebox which would be cool so I can stick with softsqueeze (the software version of squeezebox).
The flat to 30Hz Alpair5 was interesting LEANAUDIO: Report of Alpair5 Aplication but is it serious? What SPL do you need for quiet working at your PC?
foobar - can´t run it.
Hi Bob,
I´ve read your postings about foobar and I tried to install this. The idea to have dsp instead of a passive network convinced me. So I tried to run foobar on the laptop (windows 7, 64 bit) goingt out to a marantz 7.1 dsp amplifier (by using the hdmi connection) and from there direct to the speakers.
At the moment I try to bring the visaton B20 together with the dayton pt2 8 Ohm. They are installed in a baffle and a subwoofer (closed box) is added. The visatons should finally also come in a closed box.
Maybe I haven´t the right plugins installed. I can see 4 channels and I can put a filter to the channels. But only one filter to all channels. I didn´t find a way to divide the channels and put to each channel a dsp filter.
Unfortunatly I´m unversed to handle files inside a computerprogramm.
Which plugins do you use - or is there any trick to bring the dsp in line to the outgoing channels? By using the surround 5.1. function of foobar, it shows only 2 channels and it´s not possible to deal the settings.
need some help
Reinhard
Hi Bob,
I´ve read your postings about foobar and I tried to install this. The idea to have dsp instead of a passive network convinced me. So I tried to run foobar on the laptop (windows 7, 64 bit) goingt out to a marantz 7.1 dsp amplifier (by using the hdmi connection) and from there direct to the speakers.
At the moment I try to bring the visaton B20 together with the dayton pt2 8 Ohm. They are installed in a baffle and a subwoofer (closed box) is added. The visatons should finally also come in a closed box.
Maybe I haven´t the right plugins installed. I can see 4 channels and I can put a filter to the channels. But only one filter to all channels. I didn´t find a way to divide the channels and put to each channel a dsp filter.
Unfortunatly I´m unversed to handle files inside a computerprogramm.
Which plugins do you use - or is there any trick to bring the dsp in line to the outgoing channels? By using the surround 5.1. function of foobar, it shows only 2 channels and it´s not possible to deal the settings.
need some helpReinhard
Bob,
I am very encouraged by this post because it affirms what I have experienced with my own setup, which I am extremely impressed with. I am running:
Asus EeePC --> Linux --> DeadBeef player --> direct hardware output ("bit perfect") --> HotAudio DAC Wow USB (Wolfson 8524) --> Grant Fidelity B-283 (tube buffer) --> Yamaha 2-channel receiver (in Pure Direct mode) --> Dual EL70 "Castle microTowers"
As there are no passive x-over components at all, I am using Naoki Shibata's excellent 18-band "SuperEQ" DSP. Like you, I first setup an approximate mirror image of the EL70's frequency response graph and then proceeded to tweak it according to my listening preferences. I also generated a series of sine wave test tones with frequencies identical to the 18 bands of the equalizer and then used an SPL meter to fine tune each frequency to approximately the same level.
I find a "negative gain" approach to EQing works far better in this setup and this DSP makes it very easy to first get the "shape" you want and then "zero" the EQ on highest point.
I saved this "base" as a preset and then I have a couple of variations which enhance bass and/or treble slightly.
At reasonable listening levels (up to 80 or so dB) I do not find the excursion of the drivers to be excessive. Perhaps this is because the microTower enclosures do such a great job of pulling the bass out of these little EL70s that I do not feel the need to push the lower frequencies really hard.
(Note: Source files are overwhelmingly FLAC, with a bit of various quality MP3 in the mix)
I am very encouraged by this post because it affirms what I have experienced with my own setup, which I am extremely impressed with. I am running:
Asus EeePC --> Linux --> DeadBeef player --> direct hardware output ("bit perfect") --> HotAudio DAC Wow USB (Wolfson 8524) --> Grant Fidelity B-283 (tube buffer) --> Yamaha 2-channel receiver (in Pure Direct mode) --> Dual EL70 "Castle microTowers"
As there are no passive x-over components at all, I am using Naoki Shibata's excellent 18-band "SuperEQ" DSP. Like you, I first setup an approximate mirror image of the EL70's frequency response graph and then proceeded to tweak it according to my listening preferences. I also generated a series of sine wave test tones with frequencies identical to the 18 bands of the equalizer and then used an SPL meter to fine tune each frequency to approximately the same level.
I find a "negative gain" approach to EQing works far better in this setup and this DSP makes it very easy to first get the "shape" you want and then "zero" the EQ on highest point.
I saved this "base" as a preset and then I have a couple of variations which enhance bass and/or treble slightly.
At reasonable listening levels (up to 80 or so dB) I do not find the excursion of the drivers to be excessive. Perhaps this is because the microTower enclosures do such a great job of pulling the bass out of these little EL70s that I do not feel the need to push the lower frequencies really hard.
(Note: Source files are overwhelmingly FLAC, with a bit of various quality MP3 in the mix)
Last edited:
Indeed, bypassing Windows kernel mixer is critical, but ASIO is still necessary for people running Windows pre-Vista/7.
Other options to bypassing the Windows k-mixer is to bypass Windows completely. Linux offers complete control and an incredible array of powerful audio tools and DSPs.
I'm slightly confused.
I've installed Foobar in my laptop (runs out to a 2ch USB soundcard, into my system as outlined below). That's all fine.
The bit I'm confused about is the Windows mixer - I'm running Windows 7 and consider myself reasonably able to get around all the settings, but here I'm stumped. If someone could take the time to break it down simply, that would be hugely appreciated.
Could it be (under Preferences) the Output settings - you can choose which output Foobar uses, so do I just set that to the USB card?
Thanking you in advance,
Chris
I've installed Foobar in my laptop (runs out to a 2ch USB soundcard, into my system as outlined below). That's all fine.
The bit I'm confused about is the Windows mixer - I'm running Windows 7 and consider myself reasonably able to get around all the settings, but here I'm stumped. If someone could take the time to break it down simply, that would be hugely appreciated.
Could it be (under Preferences) the Output settings - you can choose which output Foobar uses, so do I just set that to the USB card?
Thanking you in advance,
Chris
Chris,
You might want to go to the Foobar2000 forum. Search around on "bit-perfect.
But to answer your question, as long as the Foobar resampler matches the bit rate and depth of your DAC, Windows 7 Direct Sound will be bit perfect. If you just can't make yourself use DS, you can use WASAPI. The only disadvantage of DS is that it will allow other sounds through the mixer. You may want that, you may not. I have all Windows sounds muted.
BTW, I upsample everything to 24/96. Of course bit-perfect goes out the window, but this allows using the Foobar volume control to pretty deep dB levels. I thint the top end sounds better when upsampled. I could be kidding myself, but that's what I hear.
Back to the OT, avoid the Foobar default EQ. It is not very good. Instead, us the foo_dsp_xgeg graphic EQ. Much better. Of course, if you have a favorite VST DSP, you can use that.
Bob
- Status
- Not open for further replies.