Jennice,
Maybe you try this cheap USB-solution: http://www.diyaudio.com/forums/showthread.php?threadid=59905
$42 is not a lot of money.
Roger
Maybe you try this cheap USB-solution: http://www.diyaudio.com/forums/showthread.php?threadid=59905
$42 is not a lot of money.
Roger
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.
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
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.
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
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
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
I suggest ...
... 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.)
....
Mercinary announcement: I work for a company that distributes M-Audio and Roland online ... firewirestuff.com / usbstuff.com / fiberstuf.com / 3dotvideo.com and more
... 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.)
....
Mercinary announcement: I work for a company that distributes M-Audio and Roland online ... firewirestuff.com / usbstuff.com / fiberstuf.com / 3dotvideo.com and more
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/
Attachments
johndiy said:
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/showthread.php?postid=1008980#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 .... .... ..
Attachments
You did OK ...
... " 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...Sections&file=index&req=viewarticle&artid=333 ... compared to your card: http://sounduk.terratec.net/modules...Sections&file=index&req=viewarticle&artid=329 )
... " 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...Sections&file=index&req=viewarticle&artid=333 ... compared to your card: http://sounduk.terratec.net/modules...Sections&file=index&req=viewarticle&artid=329 )
Re: You did OK ...
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:
lineup getting ready for a digital future, using his PC as Sound Center!
FastEddy said:
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:
lineup
getting ready for a digital future, using his PC as Sound Center!Attachments
" ... 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) ...
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 ...
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) ...
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 ...
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.
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.
" ... 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.
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.
FastEddy said:
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.
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/productpage.cfm?prodID=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 ... =
)
The differences are similar to trying to compare the quality of MP3 files to quality of WAV files ... not!
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/productpage.cfm?prodID=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 ... =
)The differences are similar to trying to compare the quality of MP3 files to quality of WAV files ... not!
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 crap, 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...
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 crap, 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...
" ... 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 ...)
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 ...)
About USB Sound Card
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
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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