| Jennice |
Hi all,
I am looking for a USB sound card, which I want to use for speaker testing (building my own speakers).
I understand that Creative's cards aren't suitable for this. Can anyone explain why?
Can anyone suggest which USB card to use, as the on-board cards usually aren't good (so people say)? (price is a factor for me :( )
Jennice |
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| sangram |
USB itself is not so great when it comes to soundcards, too much and not sufficiently predictable latency.
Actually onboard sound would be better in that respect except that its self-distortion would be unacceptably high.
You may want to consider a Live 24-bit or entry-level soundcard like a Chaintech AV-710 which is getting some decebnt reviews.
If you definitely want a USB soundcard, try one of M-Audios entry level offerings. I forget what it's called. |
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| Ouroboros |
I can thoroughly recomend the M-Audio 'Transit' USB audio interface. I use this both at home and at work for speaker and other audio transducer testing, and I 've had no problems. I use both 'True RTA' and 'Sample Champion' software.
The Transit measures well on the RMAA self-test. |
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| Jennice |
Hmm... I can't seem to find the transit card locally.
Heard about the JamLab or FastTrack devices from M-audio?
I can get those here in Denmark (DK).
---
EDIT: Seems that Pinnacle represents M-audio here in DK.
Jennice |
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| Jennice |
It's for my laptop, but I guess I could get a firewire card for it. (It has USB, so it would probably be cheaper that way), but what ideas do you guys have for FireWire cards.
Nothing exotic and excessively expensive, I hope.
One of you mentioned the Live 24bit. I can get those here, but I thought I should avoid Creative - they seem to be criticised here on diyaudio - although I don't know why. Is it their phase which is lost in it's DSP? |
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| Jennice |
| Hmm... anyone out there ? |
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| FastEddy |
" I am looking for a USB sound card, which I want to use for speaker testing (building my own speakers). ..."
Most built in sound cards simply don't have enough power / performance to drive decent sized or decent quality speakers ... unless you actually like this cute little "computer speakers" that go for $10 a set. No output power, no clarity, not decent power filtering, no
" ... I understand that Creative's cards aren't suitable for this. Can anyone explain why? ..."
IBID above.
" ... Can anyone suggest which USB card to use, as the on-board cards usually aren't good (so people say)? (price is a factor for me ) ..."
Best suggestion: a stand alone USB or FireWire interface sound "card" / audio adapter. 1) This gives you many more options than a built in sound card plug-in. 2) Upgrades are much easier when the time comes (we don't own these computers, we simply rent the technology with replacement coming every few years), 3) generally the USB or FireWire connected audio interfaces are of much better overall quality and compatibility, 4) since you need an external amplifier (pre-amp and/or combo) connectivity is easier, better, cleaner, cheaper (in many cases) ... etc.
----
disclosure: I am in the biz of selling audio interfaces and other electronics for computers, but what I say above I believe is gospel for those interested in the quality of digital to analog audio. |
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| FastEddy |
" ... I can't seem to find the transit card locally. ..."
FYI: see http://3dotaudio.com ... we ship world wide. We also happen to import from Denmark and I believe international shipping is about US$45 for the M-Audio transit ...
:smash: |
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| Jennice |
FastEddy:
Thanks for the reply. I don't plan on using the on-card amplifier.
Just the line level signal. I intend to use a seperate amplifier to drive the speaker.
The USB cards don't seem to have "power" (speaker) outputs anyway, du to the limits of the USB.
Still, I've seen people comments on the creative cards, but I can't find a precise argument on what people don't like about them.
I found the transit card in a danish shop for music instruments (mainly pro - gear), so I guess M-audio is aimed at the somewhat more "serious" user?
Jennice |
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| FastEddy |
" ... I don't plan on using the on-card amplifier. Just the line level signal. I intend to use a seperate amplifier to drive the speaker. ..."
Fine ... and I suppose you know about all the efforts needed to clean up the signal = removal of most of the PC power supply noise, etc ...? Works OK without it, but most here understand the problems and take several DIY steps to resolve this ...
The Creative stuff is as good as any, better than some. Still, depending on your PC, noise can get through from the PC switching supply in any case. It is generally better with a FireWire or USB to audio interface as there is additional filtering on most of these external interfaces as well as certain technical relationships between PC digital to USB /ÊFireWire DAC (digital to analog conversion) that works pretty good. Of course the costs of really, really good FireWire or USB DACs can be considerable .... but if you want the best quality results possible, well ...
:smash: |
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| Jennice |
I found out that the main issue with Creative cards is that they resample everything from 44.1 to 48 kHz, which causes problems for accurate measurements and phase testing.
M-audio also makes the Audiophile card, which has external PSU (not feeding from the USB power). The specs look impressive, but the cheap Transient card seems to have decent SNR specs, too.
I checked with LspCAD, and it supports 48 kHz sampling rate, which seems to be the usual for M-audio's cards.
However, it also supports just about any other imaginable rate, so I guess this leaves a free choice, if Creative's main problem is their re-sampling from 44.1 to 48 kHz.
What's this latency issue with USB devices? What does it mean? (I guess my english isn't good enough to understand). Can anyone explain?
Jennice |
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| FastEddy |
" ... at's this latency issue with USB devices? What does it mean? (I guess my english isn't good enough to understand). Can anyone explain? ..."
All electronic data transfer media, analog or digital, suffer from "latency" because of the speed of light and sound (signals) in metal conductors.
Basically, because of the nature of USB being "full duplex" (two data wires) and FireWire being "quad duplex" (four data wires), there are latency issues with USB. These have been well known for about a decade. This problem applies to all "packet switching networks" in varying degrees (USB, FireWire, modern EtherNet, SATA, etc. ... and even though modern EtherNet has eight data wires, the packet switching and handshaking overhead of EtherNet increases latency to unmanagable levels for DAC, ADC audio I/O).
Professional studio operators originally had tough sledding trying to feed USB to the control room for digital recording, then feeding digital to analog audio (DAC) back to the studio for double tracking, etc. ... especially when adding voice tracks. The musician (as listener) hears a tiny delay (the latency) and when trying to sing along (adding tracks) this latency results in "out of sync" tracks (late arriving = latency). There is no perfect solution, yet ... and the use of "simpty time codes" to square it up does not help the musician a bit.
With FireWire the latency is greatly reduced, enough so that the latency is almost undetectable by the musicians and studio operators. This is why so much of the professional studio digital to analog (DAC) and analog to digital converters are connected via FireWire (IEEE 1394a or iLink (Sony)) rather than by USB or some other scenario. It is still quite common in the professional studio to do everything as analog, first, then make the digital conversion at the final mix down .... unless the studio is completely "FireWired", then all original sources are converted to digital from the microphones and pickups directly to the digital tracks.
Important: The larger latency of USB for computer to analog audio output is not a problem for those of us looking for quality audio playback ... only. It is only of concern with bi-directional audio interfaces, analog to digital and digital to analog, (audio in & audio out), and even then there are ways to keep it all straight, unless you are doing live recording and then "double tracking".
References:
http://en.wikipedia.org/wiki/SMPTE_time_code
Compare the specs of these for clarity:
http://rolandus.com/products/produc...31&ParentId=114 ... verses ... http://rolandus.com/products/produc...04&ParentId=114 ...
http://www.m-audio.com/products/en_...ckPro-main.html ... verses ... http://www.m-audio.com/products/en_...phile-main.html ...
http://www.oxsemi.com/oxford/docume...udio/audio.html
FYI: FireWire 400 is more than twice as fast as USB 2.0 and greater bandwidth than 1000baseT "gigabit" Ethernet by about 10% ... in "bulk file transfer modes", ala music and video. FireWire 800 (1394b) is the fastest and has the smallest latency, being faster than any scenario and primarily used for video graphics rendering / networking = about 2.5 times as fast as Gigabit Ethernet. (Check out movies "Sin City" or "Sky Captain" = all FireWire 800 graphics rendering on Apple Macs. Then check out Maya software & hardware rendering scenarios.)
:smash: |
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| Jennice |
Hi FastEddy,
Thanks for your good explanation of latency.
In my case I don't think latency is an issue, as am not comparing the response (input to the sound card) to the output from the sound card.
I will be measuring the drive signal with a probe feeding the left input, and the microphone response (through a seperate preamp) on the right input channel.
I have yet to investigate if it becomes a problem during calibration, as calibration is a "loop" from output back to input. A significant delay in this chain will result in a phase issue.
Then again, as long as this calibration is aware of a phase shift, and the same applies to the real measurement, it should be ok... I think! :xeye:
I have yet to think more about this. Having to add firewire is a costly addition.
although less flexible, I have also seen manufacturers of sound cards for the PC-bus (32 bit version of the PCMCIA). I suppose bandwidth is good enough there to have little latency concerns - or am I mistaken?
Jennice |
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| FastEddy |
We always recommend the FireWire interface as a matter of course, whether output only or bi-directional input / output.
But, USB generally is a little less expensive.
As for " ... I have also seen manufacturers of sound cards for the PC-bus (32 bit version of the PCMCIA) ...", the latency problems may disappear, but the power supply noise problem then becomes an issue ... as per previous postings.
Win some, lose some, sometimes it rains. :smash: |
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| Jennice |
FastEddy,
Even though the add about giving away the PCI cards with M-audio products was a little misleading (it's for a laptop), I appreciate your feedback on this issue.
I've found a (fairly) cheap PC-card bus adaptor with USB2 and 2 x FireWire400 ports. Would that be a solution if connected to a FireWire sound card?
Jennice |
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| Jennice |
FastEddy,
I looked at some FireWire sound cards :hot:
That's expensive stuff (couldn't find M-audio products locally with FW)!
The only thing I can afford (of what I've seen) is the Behringer FCA-202.
I think I'm at the point where I have to consider if it's worth it, and dig more into the latency issue (if it's any problem for me).
Thanks for your ideas and feedback so far!
Jennice |
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| FastEddy |
... but, if you are interested in Professional, Studio Quality results, especially when it comes to dubbing, mixing tracks, adding live tracks, sophisticated multi track work, etc. ... then consider FireWire ...
Do Not Discount the abilities of the better USB sound interfaces verses built in or plug in sound cards.
Generally, digital playback is as good via the USB methodology as any other technology and usually much better than "built in" or plug in sound cards, PCI, PCMCIA (CardBus) or what have you ... Mostly because of the nature of the mass market PC and Laptop power supplies and built in or plug in PC power filtration. (PCs and Laptops all use switching power supplies, which are notoriously dirty animals = lots of noise, distortion, wheat chaff and oat husks ... :mad: )
If you want SPDIF coaxial, digital to digital output, then the external USB devices are superior to the built in and plug in cards, period. (Digital to digital output, the US$100 M-Audio Transit USB device is vastly superior to any built in or plug in sound card.)
If you want SPDIF coaxial and SPDIF optical output (Toslink, et al) then the built in and plug in cards may or may not suffice, but for reasons of cross compatibility, usefulness, versitality and resale value, again, the external USB devices are superior to any built in and plug in cards.
If you want complete, bi-directional (input & output) of digital to analog and analog to digital audio, then there is absolutely nothing wrong with the better quality external USB audio devices. (The latency question as originally addressed here is eleviated better with FireWire 1394, but this latency question does not affect good quality, standard playback and/or capture for ripping, burning or pure digital mix down ... just when mixing live analog input to previously recorded digital.)
Personally, I have a new Apple MacBook with the SPDIF optical port built in ... and it serves the purpose very well indeed, transporting 24 bit x 96k audio out to my optical port equipped playback system. (All Apple MiniMacs, iMacs, G5s, PowerBooks and the rest all are now so equipped = built in SPDIF optical ... plus FireWire ports ... plus several USB ports.)
If I were to take this Laptop into a studio for some serious mixing and/or live recording, I would take along an M-Audio FireWire devices ( http://www.m-audio.com/products/en_...phile-main.html ) or Roland FireWire device ( http://rolandus.com/products/produc...31&ParentId=114 ) ... primarily because that pesky latency question is fully addressed or at least resolved to the point on non-contention.
If I were to use this Laptop for simple, serious digital to analog (DAC) or serious SPDIF coaxial playback into a top quality solid state or tube amplified sound system, I would get the well filtered, well engineered WaveLength Audio devices USB to DAC ( http://www.wavelengthaudio.com/usbdac.html )
Thruth is one, paths are many ...
:smash: |
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| Jennice |
FastEddy,
My purpose is purely the testing of speakers. The only iussue I will have to consider is this latency, and if it makes any difference at all, if there is a delay between the cards analog output, and the sampling of the analog inputs.
I will have to dig into this issue a little more.
However, I want to thank you for all your replies to my (possibly stupid) questions about latency and similar.
I think it's time for me to sit down and think this through, and ask some questions to the software manufacurer.
Jennice |
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| FastEddy |
Consider:
If your "speakers under test" are connected to your amp, your amp to a pre-amp, your pre-amp has a (or is connected to) a SPDIF or other type of Digiral to Analog converter, your pre-amp connected DAC is then connected to your computer .... AND the sound from those speakers is recorded by your microphones to an Analog to Digital converter (same as above or other channel) ... and that analog to digital converter (A/D) is .... feed into the same computer = playback back output and bi-directional conversion D to A and then A to D back to the same computer ....
Then latency is inevitble ... and no matter what transport and/or conversion type ( methodology = USB, PCMCIA direct, plug in sound card or built in optical or FireWire ) is used ... latency will show up.
In fact latency of a very modest amount would be present whether your test system is pure analog or pure digital or a combination of the two (as is now more common practice). This is the result of the speed of sound in the air between the speakers and the microphones and the speed of light (electronic audio signals) in metal or glass conductors ... By adding all the computer processors' process / conversion times of the(internal or external) chips to the total latency ... then this may become significant enough to cause errors in your tests.
The best measurement tests of audio equipment as you are trying to do are accomplished with "pure" analog equipment = pre-amp and amp, speakers and microphones ... analog osciloscopes, spectrum analisers and signal generators.
Second best for test of speakers are "pure" analog sources played back and captured to
via bi-directional A to D and D to A converters of highest possible resolution and the Least Latency possible ... AND introducing software timing corrections to the analisys ... like spmte time code corrections in programs like "Pro Tools", etc.
Of interest: many of the better osciloscope hardware and software scenarios do a very good job of signal generation and reporting and especially display of spectrum analisys and insertion of various schemes to account for Latencies.
Of course you can use consumer and commercial and audiophile grade add-ons to your computer system, but chosing the best ways to reduce or limit Latency problems have already been described abov. My opinion backed by professionals is that FireWire interfaces for conversion of both input and output ... has the least latency problems of all.
Perdon my spelling (sic)
:smash: |
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| FastEddy |
This is a 16 bit device ... so playing 18 bit or 24 bit or whatever bit rate through it might not have the results expected ...
:( |
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| Jennice |
RG,
Thanks for your suggestion.
At the moment I don't have that much spare time, so I'll cot the corner and buy a sound card when I make up my mind. It'll leave me more time to play than build. :D
I know buying is not in the spirit of DIYaudio, but it's for speaker measurements, so the DIY aspect is not entirely lost.
Jennice |
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| RogerGustavsson |
For $42 you get a built and tested USB AD/DA.... The only thing missing is the enclosure and I believe you can even use it without one.
Roger |
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| planet10 |
Fast Eddy.
You seem to have a good handle on digitalIO devices... in terms of sound quality what FireWire DAC would you recommend (it seems most of these devices will have lots of other features that go unused)... price in mind, price no object & finally if you want to go in and tinker (ie the Metric Halo guys said they could provide the basic info i'd need to tube-i-fy their MIO). There are such a dizzying array of devices with none targeted to my need.
At the moment i have a PISMO G3 (found in the dumpster) as source, but working at an Apple dealer part-time means that a continuous variety of "free" boxes cycle thru.
dave |
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| FastEddy |
... either FireWire or USB external devices, generally, as built in sound cards and plug in sound cards have the computer power supply noise as a given = hard to filter and all computers have this problem unless special steps are taken.
* If the application is just SPDIF (digital and/or optical) input and output, The M-Audio USB Transit is hard to beat ... ~ US$100 ... mods are available from the folks at BolderCables.com and other sources.
* If the application is serious, professional, analog to digital / digital to analog (DAC) for use in a Professional Recording Studio, then there is no substitute for a decent FireWire 1394a multi channel interface then Rolandus.com , M-Audio.com , MOTU.com , DigiDesign.com , .... and several others make a very wide array of FireWire connected devices. ~US250 up thru ~US$2500 ...
* If the application is high quality digital to high quality analog output ("straight" DAC) ... then the USB interfaces discussed at length here and elsewhere should be considered. WaveLengthAudio.com springs to mind but there are dozens of folks doing very interesting work in this area ... ~US$250 thru ~US$2500 ...
* If your are just looking for a good, cheap DAC, consider at least 24 bit / 96k input / output, each channel and analog + SPDIF digital connections ... there is a plathora of used devices from Roland (Edirol) ... all solid state so the used equipment is probably not broken. Search for Edirol UA-5 and late model UA-3 types or look for 24bit/96k w/dolby decoding ... usuazlly under ~US100.
*If you can find used, quality, working FireWire I/O audio devices, expect the costs to be close to 60% of retail. (I have not seen much used FireWire as once you own it, you don't want to part with it.)
:confused:
....
Mercinary announcement: I work for a company that distributes M-Audio and Roland online ... firewirestuff.com / usbstuff.com / fiberstuf.com / 3dotvideo.com and more |
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| planet10 |
thanx... a bit more info to add to my knowledge base.
dave |
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| lineup |
:)
hello
i dont know if i did something bad
but i ordered a Terratec Universe - soundcard
eventhough they have
Firewire, which my computer supports!
I wait for my package delivery to arrive anyday.
what you think, friends
should i have chosen the firewire???
info: http://sounduk.terratec.net/
Regards and thank you for any opinion
lineup
Lineup Audio Lab
http://lineup.awardspace.com/ |
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| johndiy |
Hi Lineup
it looks great, what are you gonna use it for,audio analyser?
cheers
john |
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| lineup |
| quote: | Originally posted by johndiy
Hi Lineup
it looks great, what are you gonna use it for,audio analyser?
cheers, john |
I was thinking like this:
I have my old have EF86 tube based oscilloscope HEAVY as a fat man!
I have used this 2-channel like 5 MHz analog scope for some looking at output of my amplifiers. Sine and square waves.
i use it in final developement stage, for the trimimng with small caps, to get squarewaves not to have too much over-shooting.
But it is a drag to start up this old 100% tube scope.
Only warming up all them London english tubes, will take half an hour of time ...
Remember them vintage tube-radios!
now as i use my New super Pc so much,
I thought getting a good link into my computer and use some smart technology software
( why not best freewares like http://audacity.sourceforge.net/ )
and some software Oscilloscopes add.ons
and so get my signals into where I sit the most.
And easily be able to present my results onto my webpages:
http://lineup.awardspace.com/
I also would love a good link to PC for my DAT, Digital Audio Tape recorder machine,
16 bit 96kbps sampling, for maximum recording,
for example from my discrete transistor microphone amplifiers, like one shown in my own pictures:
http://www.diyaudio.com/forums/show...980#post1008980
Short, i have never owned a dedicated sound card - only had those built in stuff
on the motherboard.
When you are into audio you ought to give yourself something better.
Dont you think?
Regards to ellada or sidney or whereever you are right now my friend, John
from lineup
http://lineup.awardspace.com/
another good thing,
has built phono RIAA amplifier
just to attach my Grammophone and start copying my old Vinyl Long Playing 33 rpm albums
from 1965 and forward .... .... .. |
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| FastEddy |
... " 24bit S/PDIF input/output with 44.1, 48 and 96kHz " (from http://sounduk.terratec.net/ )
This is the critical spec on any sound card = the ability of any sound card to pass 24 bit / 96k audio in and out.
This sound card is as good a choice as any, certainly much better than most.
Interestingly, by using the Coaxial Digital (SPDIF) or the analog @ 24 bit 96k to connect to your sound system, the quality differences will not be detectable by your or my ears, unless the computer you have allows the internal power supply noise to "escape" through your cabling to your system (digital pre-amp input). The external "breakout box" will prbably do just fine filtering any of this off the lines.
Let us know how it goes / how it sounds.
(That Aureon 7.1 FireWire item shown above your sound card is about as good as it gets = Oxford Semiconductior chip set and good power supply filtration: http://sounduk.terratec.net/modules...ticle&artid=333 ... compared to your card: http://sounduk.terratec.net/modules...ticle&artid=329 ) |
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| lineup |
| quote: | Originally posted by FastEddy
... " 24bit S/PDIF input/output with 44.1, 48 and 96kHz " (from http://sounduk.terratec.net/ )
This is the critical spec on any sound card = the ability of any sound card to pass 24 bit / 96k audio in and out.
This sound card is as good a choice as any, certainly much better than most.
Interestingly, by using the Coaxial Digital (SPDIF) or the analog @ 24 bit 96k to connect to your sound system, the quality differences will not be detectable by your or my ears, unless the computer you have allows the internal power supply noise to "escape" through your cabling to your system (digital pre-amp input). |
thanks, FastEddy
I visited your website - so I noticed you most probably have some good knowledge :)
What I am glad about, is that there are BOTH type of digital connects on the soundcard Front Module.
SPDIF and TOS-link ( Coax and Optical ) Digital Input and Output of 24/96 digital signals.
You know there are some devices that have only one type of digital,
either Coax or Optical.
I notice some playback can be done with 24/192 but everything else is in 24/96.
I have DAT-recorder that has default 16bit/48 ksps speed.
But I can select Double Speed = 16bit/96 ksps, samples per sec.
Normal CD has got 44.1 kilosamples per second.
What I have learnt, it is not only the number of bits ( 16, 20 or 24 bits )
but also how many samples of those bits is taken per second,
that is important for good high sound quality recordings.
Aureon Universe comes with an advanced CD software interface
with most any feature.
And a remote control!
Here are the details of FrontModule 5 1/4" for put in my computer Chassi.
See also attached picture:| quote: | Front Module Connectors
* Bit-true Digital output, optical/coaxial, 44.1/48/96 kHz (TOS link)
* Bit-true Digital input, optical/coaxial, 44.1/48/96 kHz (TOS link)
* 1 Line Out, stereo (Cinch) 24 Bit/96 kHz*
* Line In, stereo (Cinch) 24 Bit/96kHz
* Phono In, stereo (Cinch)
* Microphone In, mono (6.3 mm) 24 Bit/96 kHz with Gain Control
* Headphones Out, stereo (6.3 mm) 24 Bit/96 kHz* with Volume Control
* IR Remote Control Receiver |
lineup ;) getting ready for a digital future, using his PC as Sound Center! |
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| FastEddy |
" ... What I have learnt, it is not only the number of bits ( 16, 20 or 24 bits ) but also how many samples of those bits is taken per second, that is important for good high sound quality recordings. ..."
The optimum bit rate number is 24 bits. Almost all 24 bit capable DAC / ADC devices can "interpret" 16 bit, 18 bit, 20 bit and 24 bit digital information (data packet frames) with no significant alteration to the audio content.
24 bit is commonly used for studio multi-track, mulit-channel mastering with bandwidths of 96K for each channel ! ... 192K or even 385K bandwidths are used for each encoded stereo channel (Dolby, THX, etc.) ... although confusion is the rule of the day, the manufacturers of such equipment baffelling the engineers (and customers) with their BS, hype and nominclature ...
Suffice to say that the bit rate is the most important number here: 24 bit being better. (Analog audio information is "chopped up" into bits. The larger the number of bits = the smaller the chopped up pieces and the higher the resolution.)
Bandwidth is simple, but hard to explain ... and I may generate more confusion and misinformation here as I'm not sure I fully understand every nuance of the various competing methodologies (plus Dolby and/or THX, et al):
A bandwidth of up to 96K Htz. (96,000 cycles per second), also implies, usually but not always, that the audio information has that much room or bandwidth for multichannel information. The higher the bandwidth, generally, the greater the _possible_ quality.
Many DVD movies have 48K bandwidth, most common CDs have 44K bandwidth, DVD-A audio DVD discs usually have 96K or 192K bandwidth, common FM radio transmission (broadcasts) here in the USA is around 96K bandwidth, most heavily traveled Internet Radio "broadcasts" are between 32K and 96K bandwidth with some low traffic, "higher quality" web sites "broadcasting" at 192K bandwidth.
Available bandwidth does not necessarily mean better quality. Virgin Atlantic Internet radio "broadcasts" from 32K (into USA) to 192K (into London), but virtually all of their music is CD quality or worse, 44K. (Of interest: compressed 44K CD audio can be "broadcast" in a 32 bit data frame packet) ... :bawling:
For my money I seldom purchase music CDs (44K) anymore, prefering DVD video concerts (48K to 96K) and DVD-A (or SACD) audio discs ... which my DVD players will all accomodate. My Apple MacBook laptop will pass all of these digital formats to my analog equipment, BUT the optical limitations of the Apple LossLess via optical TOSLink cable, limits the bandwidth to, apparently, 48K (?). I am in the process of obtaining a studio quality FireWire connected DAC for my audio system, knowing that whatever I play on my Apple internal DVD/CD player will be passed to my analog system at full 24 bit/96K or better bandwidth directly via analog cables ... these cables will be analog, balanced XLR types, not digital coaxial or optical, with considerable attention being made to the power supply filters for the Apple andthe DAC/ADC ...
:confused: |
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| dangus |
You're mixing bitrate and sample rate. Sample rate is how many digital samples are recorded per second, and thanks to pesky laws of mathematics, the maximum audio bandwidth is slightly less than half the sample rate. So a 48k samples/second rate means that the audio cuts off just below 24 kHz.
Bitrate means the amount of data used per second. PCM audio (CD quality) is about 1.4 megabits per second (16 bits * 44 k samples/second * 2 channels). Compressed formats (MP3, Dolby Digital etc) reduce the bitrate while keeping the original sample rate and bit depth (resolution). Dolby Digital manages to cram 5 full-range channels plus the limited bandwidth LFE into something like 384k to 448k bits per second, less than a third that of CD audio. |
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| FastEddy |
" ... You're mixing bitrate and sample rate. Sample rate is how many digital samples are recorded per second ..."
This is not something that can be debated: 24 bit data frames of analog information are not sampled at 24 bits per second ... or 24 thousand bits per second (or any other number of "bits per second.)
The sample rate for 24 bit / 96K audio is 96,000 times per second, each sample is divided into (digitally converted to or from) 24 bit data ... 24 bit data frames.
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| Schaef |
| quote: | Originally posted by FastEddy
" ... You're mixing bitrate and sample rate. Sample rate is how many digital samples are recorded per second ..."
This is not something that can be debated: 24 bit data frames of analog information are not sampled at 24 bits per second ... or 24 thousand bits per second (or any other number of "bits per second.)
The sample rate for 24 bit / 96K audio is 96,000 times per second, each sample is divided into (digitally converted to or from) 24 bit data ... 24 bit data frames.
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And if you were transmitting the 48 bits (24 per stereo channel) in parallel, then yes, the bit rate would be 96,000. However, SPDIF, TOSLINK and others transmit data serially, one bit after another, so in order to achieve that same transmission, the bitrate has to be increased to 48 * 96,000 per second.
Dangus was merely pointing out that the 384k you quote for things like Dolby Digital is a compressed bit rate, not a sample rate. I believe DD is actually a 16 bit/48kHz sample rate after de-compression, but I'm not positive. The 384k is merely how quickly it transmits the compressed bitstream.
So, sample rate is one thing, namely 24/96k, or how fast and deep samples are taken per second. The bitrate is how fast those bits are sent from the ADC (or to the DAC) to the rest of the equipment and is at a much higher rate. In other words, even though the samples are 24/96k, the actual physical transmission down the cable (or fiber) is much higher. |
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| FastEddy |
Schaef: Thanks for the clarification ... You are of course quite correct ...
The original point I was trying to make several posts back was that the various manufacturers' marketing types and some engineers are mis-leading a lot of folks (including me) about their actual performance specifications.
Generally I consider only those devices that are "24bit/96K" or better, leaving the "lower resolution" devices out of the discussion because of this (sometimes deliberate) confusion.
An interesting example that I just discovered: http://www.samsontech.com/products/...=1810&brandID=2 ... a "new" USB connected microphone that proports to be "studio quality". Its problem: " * 16-bit sample resolution ... * Supports 8 kHz, 11.025 kHz, 22.05 kHz, 44.1 kHz, and 48 kHz sampling rates ... "
Even considering that this is a single channel device, it is a long way from what I would consider to be "studio quality" ... if the A to D processor is not converting the analog signal using a 24 bit math / data frame then I don't give it much credibility. (Likewise, if any digital output devices don't have the ability to translate multi-channel 24 bit audio back to analog ... = :apathic: )
The differences are similar to trying to compare the quality of MP3 files to quality of WAV files ... not! |
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| Schaef |
I absolutely agree that the marketing of computer audio equipment leaves a lot to be desired. There is so much misleading information in ads these days its virtually impossible to tell what the device is truly capable of.
Creative is a good example of this, they had a sound card that was advertised as 24/96 capable. When you finally got through all of the ****, you found out that it could do its calculations on the board at 24/96 but only output 16/48! I would also agree that a "studio" mic recording at 16/48 isn't really studio. Although, the mic itself may actually be studio quality, its just the interface that isn't.
Also the next time some congress critter or legal type goes on and on about mp3's being "perfect digital copies" I'm going to slap them!!! I don't know how many times I've had to explain to people that mp3 is a lossy compression algorithm and there is a loss of quality with it.
Oh well, off to other things now... |
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| FastEddy |
" ... the next time some congress critter or legal type goes on and on about mp3's being "perfect digital copies" I'm going to slap them!!! ..."
Probably aught to do this, regularily, just on general principles, or whenever they open their mouths without engaging their brains. (I go to town hall meetings directly from baseball practice, fully equipped ...)
:smash: |
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| cdfr |
Hi,
I was wondering if anybody had tried the EMU 0404 USB sound card. On paper it seems to be quite descent 24/192 AKM, class A preamp...
Balanced input and output (which I want to use)
I want to use it to digitize my LPs, Reel to Reel, tapes etc..
Of course I plan to use it as a D/A for the PC on my stereo and also as an SPDIF input. On that topic I had read in the forum that Creative sound card forced the resampling of any spdif input, I am curious to know if this is unfortunately also the case on their semi-professional stuff like the EMU 0404 USB.
Any input would be valuable,
Thanks,
cdfr |
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| FastEddy |
" ... offering pristine 24-bit/192kHz recording and playback ..." using " ... Premium 24-bit/192kHz A/D and D/A converters ..." From: http://www.emu.com/products/product...1&product=15186
The E-MU 0404 USB2.0 bi-directional audio interface ("sound card").
I wasn't aware that it was possible to do a USB audio interface that would work in both directions (rip & roar) and maintain the 24bit/192k data path ... I knew it was possible in "half-duplex", one way at a time, but was lead to believe that the handshaking and jitter would overwhelm the 24bit/192k bandwidth ... The E-MU folks must be using some kind of propriatary streaming technic as they appear to be the only ones making anything like this.
Hats off to 'em if it works and works well. Please let us all know how it works !!
:cool:
(Caution: I will bet you should have genuine USB 2.0 ports on your computer, tied directly to the 2.5 and up PCI bus, no USB hubs in circuit, and a USB2.0 cable length shorter that 3 meters / 10 feet, directly connected to the port. Also I would recommend not having too many other high speed USB gadgets plugged into the computer. A 24bit/192k data stream is almost as fat as a quality DV camera's data stream = so similar considerations apply. Also note the cautionary message at bottom of their page: "Macintosh analog operation up to 96kHz and digital operation up to 48kHz only at this time - check www.emu.com for updates." ... that's cause most Macs use the USB circuits for all keyboards, mice, joysticks, printers, etc., and they don't have the internal 64bit PCI bus that some modern PCs have (yet).) |
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| Jennice |
If I'm forrectly informed, E-MU is owned by Creative, and technology sharing would be obvious. I just hope (for E-mu's reputation) that their studio gear doesn't become over-spec'ed like most PC equipment is.
As for the microphone someone commented on earlier:
Until 24/96 was invented, studios were happily using 16 bit as "studio quality". When playing back a CD, it's 16 bit, too.
I my oppinion, 16 bit can be studio quality. To me, tt's all about noise floor and dynamic range, really. A lot of "personal use" (and some pro-stuff) 24 bit equipment doesn't have a corresponding noise floor anyway, so effectively you don't get the number of signal level steps suggested by 24 bit equipment anyway. |
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| planet10 |
| quote: | Originally posted by Jennice
16 bit can be studio quality. To me, tt's all about noise floor and dynamic range, really. A lot of "personal use" (and some pro-stuff) 24 bit equipment doesn't have a corresponding noise floor anyway, so effectively you don't get the number of signal level steps suggested by 24 bit equipment anyway. |
Anything beyond 21-22 bits is probably irrelevant except when you are doing math on the signal -- then you better have 32 bit+ ... what is important is the sampling rate. with the 1st converters starting to hit 384kHz we are just starting to get into the range where digital can really start to compete with analog. (BTW, when i 1st read Sony's CD white paper a couple years before the 1st CD players shipped (1979?), i was taking an advanced statistics course on sampling theory. At the time i said that they needed to get sampling rates up at least 8x 44 khz before they could compete -- nothing i've heard or seen has changed my mind on that statement based solely on the math). I find it quite amazing that they have managed to make CD players that sound as good as a good one does.
dave |
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| AndrewT |
Hi Planet,
considering that the CD standard is designed to use 1976 computer technology, I too am amazed they eventually got it to sound so good.
Shame on the industry for pushing lossy compression standards onto the download market, which appears to be taking over. Woh betide us for letting this happen.
If they had gone to 192Ksampling and 20bits then lossy compression might have been acceptable.
Even 5.1 audio off DVDvideo cannot meet that standard (because they don't care to try).
Based on your long experience, just how far short is 192ksampling? |
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| FastEddy |
Jennice: " ... [IMOP] 16 bit can be studio quality. To me, it's all about noise floor and dynamic range, really. A lot of "personal use" (and some pro-stuff) 24 bit equipment doesn't have a corresponding noise floor anyway, so effectively you don't get the number of signal level steps suggested by 24 bit equipment anywayÊ..."
planet10: " ... Anything beyond 21-22 bits is probably irrelevant except when you are doing math on the signal -- then you better have 32 bit+ ... what is important is the sampling rate. ..."
Of course there is the DVD video movie sound tracks = all 24bit (48k to 96k) ... and all DVD-A and most SACD is 24bit (48k to 192k).
The reduction from 24bit back to 16bit conversion often loses something in the translation = depending on the math / firmware of the D to D or D to A converters.
No experienced recording studio engineer would use a 16bit data path for the master "tapes".
FYI:
16 bit has dynamic range ~ 88 db bandwidth, max. = between the hush of a crowded auditorium (+ 40 to 50 dba) and the blast of a stack of cranked up Marshals (+95 to +100 dba) ...
24bit has the dynamic ~ 110 db to 120 db bandwidth = between the hush of a pine forest (+ 2 to +3 dba) and the blast of an F-16 @ 100 meters (+ 125 dba) ... Threshold of pain ~= +130 dba.
Recording & playback devices:
http://rolandus.com/products/produc...02&ParentId=114
http://www.m-audio.com/products/en_...re410-main.html
http://www.m-audio.com/products/en_...e1814-main.html
http://industrialcomponent.com/maudio/lightbridge.html
http://www.digitalaudio.dk/ax24.htm (noise floor >> 110 db down)
Playback devices:
http://www.m-audio.com/products/en_...phile-main.html (claimed noise floor >> 100 db down)
http://www.m-audio.com/products/en_...nsit-focus.html
http://www.emu.com/products/product...1&product=15185
http://oppodigital.com/dv981hd/dv981hd_index.html
A content benchmark:
Compare Muddy Waters' album "Folksinger", 16bit CD verses 24bit DVD-A. Noticably better at 24bit DVD-A (almost as good as the recently rereleased vinyl) =
16bit CD = http://www.amazon.com/Folk-Singer-M...ie=UTF8&s=music
24bit DVD-A = http://classicrecords.com/catalog/s...m?sku=HDAD-2008
The above CD when compared to the DVD-A disc = "All CDs suck" - Bob Dylan
(Mercinary announcement: If anyone wants to add FireWire 1394 to a system to take advantage of the fatter bandwidth & multichannel 24bit I/O, see: http://industrialcomponent.com/fire...f/fws46603.html ... throw the 4-pin card away (Mac or Linux only, non-OHCI junk), keep the 6-pin card & cable (OHCI class = ASIO compatible, no driver required, Mac, Linux or PC) ... all for the cost of the cable! ... and my company is dealer/distributor for all above except the DAD, the E-MU "sound cards" and the OPPO Digital player.) |
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| Jennice |
Eddy,
I know what you mean about the dynamic range, but when squeezing the F16 into your living room, you loose the quietness of the pine forrest into the noise floor in most cases.
High level signals are easier to handle in terms of noise floor, but when your signal level is about 1 - 2 volts, your noise floor better be below 2 microvolts, if you want to enjoy the 120 dymanic range. Is it really down there?
Papers are easy to deal with. Real life can be harder. Consider this.
And please do keep your commercial announcements away from this forum! This is not a trade fair.
Jennice |
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| planet10 |
| quote: | Originally posted by AndrewT
Based on your long experience, just how far short is 192ksampling? |
I can only speak from theory... alot closer than 44 kHz. Just like when doing a 1st order XO in a speaker, each driver needs to reach at least 2 octaves beyond the XO point. If we look at digital in the same way... with a CD we have 0 octaves, 192 is 2 octaves... and that is if we assume that 20 kHz is enuff (and there are some very valid thought experiments that show that more is better.
dave |
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| planet10 |
| quote: | Originally posted by FastEddy
Of course there is the DVD video movie sound tracks = all 24bit (48k to 96k) ... and all DVD-A and most SACD is 24bit (48k to 192k). |
They have room for 24 bit, but the effective resolution is actually less... at least with today's tech. 24 should give us enuff bit depth, sample rate just needs to get up. It is that German firm you pointed out that is pioneering the 384 kHz sampling.
dave |
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| AndrewT |
Hi,
I have read that DVDvideo does not use all it's audio capability for the audio channels.| quote: | | Of course there is the DVD video movie sound tracks = all 24bit (48k to 96k) . |
I have seen quoted figures of 18bit to 20bit as the most common.
If I were to speculate, I suspect the lesser channels resort to 18bit 48kHz to save bandwidth for much of the time and only switch to higher resolution when the producer thinks we need it.
Then they go and apply lossy compression and we end up with little better than high compression MP3 quality for most of the audio content.
The producers give us what they think we will put up with.
DVB is even worse, particularly on the commercial channels where the advertisers pay for their bit of bandwidth. Now there's an experiment; compare audio quality between adverts and piped music.
Maybe we should lobby for more audio quality at the expense of the "extra features" they attract us with. |
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| richie00boy |
| I'd like to see better video quality first -- on some shows the picture quality is shocking. |
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| planet10 |
| quote: | Originally posted by richie00boy
I'd like to see better video quality first -- on some shows the picture quality is shocking. |
With audio we are talking about 1976 computer tech, with the actual picture, we pretty much are still using early 1950s technology. The big leap in video is HD. The cost of flat panel screens are plummetting and as they do their price point meets more and more peoples need to have a new TV, and a mandated obselence (at least in NA) is legislated to happen real soon.
These new TVs (ATM) still have to limp along with upsampled DVD or you get limited HD from your cable provider (or more -- i understand) from some of the sat vendors. It is still at least a few years before we will be able to get our video on BluRay or HD-DVD (damn them for 2 standards again -- if they don't get that sorted the guys like Apple & iTunes + the cable & staelitte guys will establish a download model that makes having a silver disc a who cares.
dave |
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| FastEddy |
" ... we will be able to get our video on BluRay or HD-DVD ... damn them for 2 standards again -- if they don't get that sorted the guys like Apple & iTunes + the cable & staelitte guys will establish a download model ..."
You got that right ... and sure as heck they will screw it up. (HD Radio = no better than Virgin Atlantic on line = 32k bps = :>( ... Apple "Lost Less, found less" optical audio = no better than 16 bit = :>( ... )
BluRay sucks already = no significant content available = no decent movies yet = no "pure" audio at all that I can find. The only thing going for BluRay is a slightly wider audio bandwidth.
HD DVD ... Look for players that can handle 1080i = maximum required for this.
Flat Panel monitors / TVs of the 16x9 viewing ratio (aspect) are all that is needed and these are (as you say) dropping fast in price. (I just got a couple of flat screens for the office, specifying 16X9 / 20 to 21 inch screen ... and I have a 16X9 24" Samsung at home ... but I don't think I'll be getting any 18.5X9 or what ever they are calling it as these are not coming down in price at all and so far have limited models to chose from = all too big for me, the office or the wife.)
DVD-A ... seems to be as good or better than SACD and is playable on all DVD players ... you just have to search for a decent "universal" player with 24bit/96k or better DAC. (And you can burn your own at home without any special equipment, licenses, decoders or encoders ... other than a DVD-RW burner/player.)
Sony may be biting the big on ... again ... as BetaMax, SACD and BluRay just don't have the mass of content that VHS, DVD-A and HD DVD have already ... Sony also seems to be shooting themselves in the foot with their own content. That "Modern Times" / Bob Dylan album is (deliberately?) a perfect screw up of a decent set of tunes by over compression of the studio produce, almost as if one division of Sony wasn't aware that the other division was actively promoting high resolution audio (SACD) = :apathic: |
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| richie00boy |
Dave, I'm on about some (not all) shows that seem to be MPEG-1 quality at best, with horrendous pixellation on contrast changes. These tend to be shows from the US I find. Not all US shows I hasten to add, just some.
Whan I'm seeing the standard of LCD TV and what not. They need to sort that out before we get excited about HD capable TV sets. |
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| jleaman |
As i have just been in school for the past week and a half, my teacher has just given me the opportunity to buy some tools that will help me in the best way, with designing speakers and with placements and other sorts of things. As this course is designed for me to start installing major home theater systems on Vancouver island, i now see that having this course was well worth the money.
He has recommended the Following.
Emu 0404 USb sound card $200.00
Behringer ECM 8000 Microphone $60$
and since i was in school i get a discount for software for RTA SPL and room placement and other very useful tools that i need to have Truaudio.com for RTA level software.
Jase |
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| rfenergy |
Hello,
I guess this thread is really old in internet time , but here is a great book explaining quite a bit (no pun intended) on digital audio.
Digital Audio Explained by Nika Aldrich
http://www.cadenzarecording.com/
I am not connected to the book or website.
I do know Nika from rec audio pro forums and gearslutz forums.
Good read for anyone interested in the fundamentals of digital audio.
Out,
Robert |
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| FastEddy |
"Digital Audio Explained" by Nika Aldrich
Looking at the web site ... my impression is that this is very well thought out, but not for the faint at heart (pro engineer only) ...
Anyone got anything that might appeal to musicians? (Definition: Musician: ... can do, but doesn't really know how it happens. ... an artist totally clueless about what happens after the noise hits the microphone ...)
Even the wikipedia entries about "loudness wars", "digital audio", etc ... don't quite boil it down to the musician's level. "Hey man, I know what I like ... What's a bit? and how many do I need?" |
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| yangsmm |
http://www.digihobby.co.kr/front/ph...display_group=1
Ester-200 have 4 buttons to control winamp and etc.
It also have two usb ports and headphone jack. It has 2-watt power amp inside for speakers.
If you want to use another preamp, then Ester-200 support Optical and Coaxial Digital Output. |
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