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Topic: Analog Line in connection to e-mu 0404 USB? (Read 11583 times) previous topic - next topic
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Analog Line in connection to e-mu 0404 USB?

Hello all,

Apologies in advance if this is simple or silly question -

Based on my research, I have made up my mind to get the E-mu 0404 USB but before I place the order, I had one hesitation and wanted to clear it up by asking on this forum...Here goes -

Essentially, I want to connect my Nakamichi Dragon tape deck to capture some of my live recordings in 24bit/192khz...so essentially I will have two RCA outputs from my Dragon and I am confused about the Mic/Hi-Z/Line inputs on the 0404...I have never used these Neutrik combo jacks and want to make sure I have it right....

Guess my basic question is - How would I connect the 2 RCA outputs from the NAK to the 0404...Is it as simple as buying a RCA (2 female jacks for each channel) to 1/4inch stereo (male) adapter that you connect to the RCA inputs and insert the 1/4 inch in one of the 0404 inputs?

If that is in fact the case, then my next question is - There are two inputs marked A and B (for Mic/Hi-Z/Line) and I was wondering are each of those in fact self sustaining stereo inputs or each represents a mono channel? Guess where I am going is - If they are mono then I need to route my Left channel RCA (from the deck) to input marked A and the Right channel RCA to the one labelled B...However, if each is a stereo input then I only have to use a single input (either A or B)...

Hope my rambling made sense.

Thanks so much in advance...

Rajiv

Analog Line in connection to e-mu 0404 USB?

Reply #1 –
Each Emu input is one channel, so tape out L goes to one, tape out R to the other.

The cables you need are RCA to 1/4 TS. You cannot get balanced inputs if you don’t have balanced outputs. Besides, balanced would gain you nothing with a line level connection. The plug goes into the center of the Emu input connector.

Analog Line in connection to e-mu 0404 USB?

Reply #2 –
Quote from: AndyH-ha on
Each Emu input is one channel, so tape out L goes to one, tape out R to the other.

The cables you need are RCA to 1/4 TS. You cannot get balanced inputs if you don’t have balanced outputs. Besides, balanced would gain you nothing with a line level connection. The plug goes into the center of the Emu input connector.

Thanks so much for the response...Glad I asked as I was unaware that each was a mono channel and so I will have to use both A and B to route my L and R channels.

That said, not sure about the comment you made about balanced inputs...may I ask what the difference between balanced and unbalanced mean and what kind of out will I get from my deck...

Also, any specific kinds of RCA to 1/4 TS adapter? Is it your garden variety RCA to 1/4 or is there something different? Maybe if you can post a link of the exact adapter needed I can verify and be sure.

Thanks again
R

Analog Line in connection to e-mu 0404 USB?

Reply #3 –
There is nothing special about the plug, just 1/4 " phone, also know as TS or mono.

Balanced lines are mostly used for microphones, from the microphone to the microphone preamp. The microphone has to produce a balanced output and the preamp must be designed to input a balanced signal. The signal, split along both sides of the circuit at the microphone, adds back together at the balanced input while any hum or interference picked up along the way cancels itself out.

Since the microphone signal can be very low level, any outside noise induced in the cable can noticeably degrade signal to noise. This can be important in studios or live performances where the microphone might be on a long cable. There might be many other cables and power lines running around, each producing its own varying field that could induce into the microphone cable. In professional equipment, all the lines might be balanced, just for extra insurance, but beyond the microphone preamp there is much less benefit.

The tape deck outputs are line level. This is a much higher voltage. RCA outputs are not balanced (the circuit behind the jacks must be balanced and three independent wires are required) so there is nothing to work with at the balanced input on the soundcard. The cable from deck to the USB unit is likely to be fairly short, but people run 50 feet or more without problems. Ordinary shielded cable works fine as long as you don't parallel them with mains power lines.

Analog Line in connection to e-mu 0404 USB?

Reply #4 –
Quote from: pandeyrajiv on
24bit/192khz

Why 24/192?

16/44.1 or 16/48 (depending on the application) should me more than adequate for the source material, at least as a final delivery format after processing.

Analog Line in connection to e-mu 0404 USB?

Reply #5 –
Quote from: AndyH-ha on
There is nothing special about the plug, just 1/4 " phone, also know as TS or mono.

Balanced lines are mostly used for microphones, from the microphone to the microphone preamp. The microphone has to produce a balanced output and the preamp must be designed to input a balanced signal. The signal, split along both sides of the circuit at the microphone, adds back together at the balanced input while any hum or interference picked up along the way cancels itself out.

Since the microphone signal can be very low level, any outside noise induced in the cable can noticeably degrade signal to noise. This can be important in studios or live performances where the microphone might be on a long cable. There might be many other cables and power lines running around, each producing its own varying field that could induce into the microphone cable. In professional equipment, all the lines might be balanced, just for extra insurance, but beyond the microphone preamp there is much less benefit.

The tape deck outputs are line level. This is a much higher voltage. RCA outputs are not balanced (the circuit behind the jacks must be balanced and three independent wires are required) so there is nothing to work with at the balanced input on the soundcard. The cable from deck to the USB unit is likely to be fairly short, but people run 50 feet or more without problems. Ordinary shielded cable works fine as long as you don't parallel them with mains power lines.


Thanks so much for that explanation and that ties with the issue of live recording I used to do without a mic-preamp...I would end up overdriving my mic input on my recorder and get distortion...your explanation clears some aspects of that (not the distortion but the principle).

And thanks for clarifying the RCA output...so I guess in the end it is a simple *Two* "Single female RCA's to a 1/4 inch TS(or mono jack) and each feeding the input A and B respectively.

Awrighty then! Off I go and place my order...

R

Analog Line in connection to e-mu 0404 USB?

Reply #6 –
Quote from: greynol on
Quote from: pandeyrajiv on
24bit/192khz

Why 24/192?

16/44.1 or 16/48 (depending on the application) should me more than adequate for the source material, at least as a final delivery format after processing.


I understand what you are saying and it may seem like overkill but I am trying to digitize my music in the highest possible bitrate in hopes a little bit (even if its a bit more) can be captured with the higher bitrate...These are rare shows I recorded and so the higher the better...

Analog Line in connection to e-mu 0404 USB?

Reply #7 –
Quote from: pandeyrajiv on
in hopes a little bit (even if its a bit more) can be captured with the higher bitrate

Unless your hardware is defective at the lower settings you will gain nothing.  There is no way that your source tapes will come close to providing the SNR capable of 16-bit audio.  If your tapes produce frequencies much above 20kHz that isn't simply noise (highly doubtful), you won't be able to hear them anyway.

Analog Line in connection to e-mu 0404 USB?

Reply #8 –
Quote from: greynol on
Quote from: pandeyrajiv on
in hopes a little bit (even if its a bit more) can be captured with the higher bitrate

it won't.

Care to explain to me why not....Years ago, I did this very experiment and captured audio in 16bit/44.1 and then 16bit/48khz and then did a difference between them...I forget the exact details off the top of my head but I did it in  Cool Edit Pro where you inserted the inverse of one wav file over the other and if they were identical they should cancel each other out but they did not and there was a bit of that difference left over in the form of a faint sound wave - almost sounded like a thin scrape from the top (analogous to skimming cream off the top of milk - most is left behind but some is scraped off) ...

Aside from this little experiment, from a physics standpoint and my calculus days in college, isn't all digital capture an attempt to break an analog waveform into slices and combining them together to get a waveform that is as close to analog as possible and the smaller (or more) the slices the more closer the waveform to the original analog waveform with no jagged edges?

As a sidenote, if there is no difference between the 16/44.1 and a 24 bit/192 then why do my ears hear a difference when I play back some of the higher res files I have...an example is David Crosby's If I could only remember my name that was remastered by Stephen Barncard and released as an audio CD and also as a DVD-Audio with 24/96 5 ch mix and also a 24 bit/192 Khz 2 ch version...When I play the 24/192 side by side to the 16/44.1 I can hear a definite difference (I am decoding both versions saved as FLAC's on my Popcorn Hour C-200 fed to my Onkyo reciever 805s via HDMI) 

I also got a vinyl capture of a 180g LP transfer of Miles Davis' Bitches Brew that was released recently at its 40th anniversary...there were two captures - the 16/44 and the 24/192 version and again I can hear the difference in sound...

R

Analog Line in connection to e-mu 0404 USB?

Reply #9 –
Quote from: pandeyrajiv on
Aside from this little experiment, from a physics standpoint and my calculus days in college, isn't all digital capture an attempt to break an analog waveform into slices and combining them together to get a waveform that is as close to analog as possible and the smaller (or more) the slices the more closer the waveform to the original analog waveform with no jagged edges?

A proper reconstruction filter does not produce jagged edges.

Regarding your tests, (1) any claimed differences in sound must be supported with objective evidence and (2) you need to make sure you are comparing two versions from the same master otherwise we're talking apples and oranges.

Please review our terms of service, to which you agreed upon registering, paying special attention to #8:
http://www.hydrogenaudio.org/forums/index.php?showtopic=3974

Analog Line in connection to e-mu 0404 USB?

Reply #10 –
Quote from: greynol on
Quote from: pandeyrajiv on
Aside from this little experiment, from a physics standpoint and my calculus days in college, isn't all digital capture an attempt to break an analog waveform into slices and combining them together to get a waveform that is as close to analog as possible and the smaller (or more) the slices the more closer the waveform to the original analog waveform with no jagged edges?

A proper reconstruction filter does not produce jagged edges.

Regarding your tests, (1) any claimed differences in sound must be supported with objective evidence and (2) you need to make sure you are comparing two versions from the same master otherwise we're talking apples and oranges.

Please review our terms of service, to which you agreed upon registering, paying special attention to #8:
http://www.hydrogenaudio.org/forums/index.php?showtopic=3974

I understand what you are saying and unfortunately I do not have the hard evidence or facts to present and do not have the time to re-do them to prove my point...whatever I am saying is subjective...All I can say is listen to them yourself...

But that said, my question to you is - If what you say is true that tapes do have a SNR to even support 16bit audio then why did Sony invent the SBM adapter where there were processing in 24 bit and then dithering down to 16 bit as output...and why are studios trying to remaster older albums using higher bitrates? Is this all an attempt at making a sucker of the consumer?


Re: the apples and apples compare - Well, the Miles Davis capture was done in 24 bit/192 first and then dithered down to 16/44.1 and my ears (again, its subjective maybe) can feel more warmth in the 24/192 track over the dithered 16/44 track...

Analog Line in connection to e-mu 0404 USB?

Reply #11 –
Quote from: pandeyrajiv on
Is this all an attempt at making a sucker of the consumer?

Pretty much, yes.

Anyway, I just though I'd let you know that keeping massive files will probably gain you nothing beyond peace of mind.

Analog Line in connection to e-mu 0404 USB?

Reply #12 –
Quote from: greynol on
Quote from: pandeyrajiv on
Is this all an attempt at making a sucker of the consumer?

Pretty much, yes.

Allright since you are the moderator and I am just a newbie here, let me ask you for facts to back your claim as well...Can you conclusively prove to me with hard evidence where there are sample files captured in 16/44 and then in something higher like 16/48(or 24/96 etc) and show evidence that there was nothing gained? Is there some location you can point me to which shows the hard facts via test results and also sample files that I can download and play around with???

I am not trying to be a smarta## here but if you as a moderator are so sure that nothing is gained by this extra bitrate I want to be convinced by hard facts as well (as per your TOS #8 that you pointed me to) and you just might save me some money and you will have my thanks!

Analog Line in connection to e-mu 0404 USB?

Reply #13 –
It doesn't work that way.  TOS #8 is about demonstrating difference; not proving that differences cannot exist.

You can choose whatever settings you like, just don't come here and tell us that one sounds better than the other without objective evidence.

My insistence that you won't be able to tell the difference is based on the relative signal to noise ratio and frequency response capabilities of 44.1/16, cassette playback and limits of human hearing.  This has all been discussed to death on this forum.  If you're interested in more information, feel free to search the site.

Analog Line in connection to e-mu 0404 USB?

Reply #14 –
Quote
...I did it in Cool Edit Pro where you inserted the inverse of one wav file over the other and if they were identical they should cancel each other out but they did not and there was a bit of that difference left over in the form of a faint sound wave...
  A couple of weeks ago there was a post about subtracting an MP3 from the original to hear the "difference".  I came-up with a few quick experiments (that you can do with Cool Edit or any audio editor) to show how two identical sounding files can have dramatic differences when mathematically subtracted.   

In these experiments, the sound of the subtracted file has no relationship the the difference that you hear (or don't hear) when the files are played separately.  (Click here[/u])

Analog Line in connection to e-mu 0404 USB?

Reply #15 –
Quote from: greynol on
16/44.1 or 16/48 (depending on the application) should me more than adequate for the source material, at least as a final delivery format after processing.
Yes, If he plans to do some processing he should use higher bit-depths and/or higher sampling rates. Some plug-ins are still crappy with 44.1 or 48 kHz, introducing their own aliasing artifacts. For old tapes 16/44.1 really should be enough for the final delivery format.


Quote from: pandeyrajiv on
Quote from: greynol on
Quote from: pandeyrajiv on
Is this all an attempt at making a sucker of the consumer?

Pretty much, yes.

Allright since you are the moderator and I am just a newbie here, let me ask you for facts to back your claim as well...Can you conclusively prove to me with hard evidence where there are sample files captured in 16/44 and then in something higher like 16/48(or 24/96 etc) and show evidence that there was nothing gained? Is there some location you can point me to which shows the hard facts via test results and also sample files that I can download and play around with???

I am not trying to be a smarta## here but if you as a moderator are so sure that nothing is gained by this extra bitrate I want to be convinced by hard facts as well (as per your TOS #8 that you pointed me to) and you just might save me some money and you will have my thanks!
Greynol pretty much explained how this works. Here on hydrogenaudio something that attempts to challenge established findings - and that 16/44.1 are more than enough for good audio quality is something like that - has to be proven by facts. Countless studies from 30, 40, 50 or 60 years ago have proven that 16/44.1 is sufficient for high quality audio. And with todays technology, even 16 bit resolution can contain higher resolution themselve, only by use of clever dithering or noiseshaping techniques. I´m a staunch believer in 24/96 myself - but I´m unable to prove it. That´s a case where evidence is clearly speaking against me. Search the site for discussions about 24/96, DVD-Audio or SACD - every test result (from DBT or otherwise) is at best inconclusive when it comes to the actual audibility of such high data rates. I have to accept that I may be a victim of a placebo effect - and so should you. Be careful with statements about superiority.

Also, have you ever thought about how your live recordings were engineered? You recorded them, yes. But how? Did you record the feed out of the mixing console before it was treated by an equalizer, a reverb processor or something else? Or did you use a microphone? In any case, there won´t be much material above 20 kHz that could be that important. And then you´d have constant noise during a live concert - for that even 12 or 8 bit could be good enough with good dithering. Furthermore, are you sure that your Nakamichi Dragon recorded something above 20 kHz? Signals around these frequencies are usually quite low in volume so I wouldn´t be surprised if they would drown in the tape noisefloor. To my knowledge there were only few tapedecks around which could actually record something above 20 kHz - and they came from Pioneer and used Dolby S. I´m just guessing of course. Correct me about those things.

Regarding the craze about remasters: Greynol is 95% right. Most remasters are only louder nowadays. Or some equalizing was applied to the material in order to adapt the sonics to todays taste. Sadly, there are very few remasters which earn that title, remasters where the remastering engineer did use a lot of care to configure the heads for the tape playback machine for example. Or when old multitrack recordings receive a completely new downmix (this happens with soundtracks sometimes). Or when errors are removed. But these remasters are seldomly done today, it´s just too expensive when you can apply some equalizing and put a sticker on the CD cover afterwards that claims "remastered".
marlene-d.blogspot.com

Analog Line in connection to e-mu 0404 USB?

Reply #16 –
The mix together of an inverted and a non-inverted file is done sample by sample. If the files are identical, the net sum is zero. However, offset one of the files by only one sample and the result is meaningless. Some samples may sum to zero, very likely some samples will have a greater absolute value than either source.

There are two aspects to "bitrate," the sample rate and the bit depth. Any given sample rate (44.1kHz, 48kHz, etc.) captures every frequency up to ½ the sample rate. Nothing below that frequency cutoff is lost, that is basic physics. Actually the upper frequency approaches ½ the sample rate as a limit, there are some practical considerations such as necessary filters that lower it just a little. Your cassettes will not contain any music signals at a higher frequency than 44.1kHz can capture, aside from the fact that human hearing doesn't work there anyway.

There are two aspects of bit depth. The first is the absolute dynamic range, roughly 6dB per bit. Thus for sixteen bits, about 96dB. This is definitely more than cassettes can support. The cassette noise floor is likely to be no lower than about -65dB, meaning there is little but noise to capture between -96dB and -65dB. Using 24 bit just means finer distinctions of the very low level noise.

The second bit depth aspect is level gradations, and the associated quantization error/noise. Going from 8 bit, with its 256 amplitude levels to 16 bit with 65,536 amplitude levels might make for a somewhat smoother sounding result. It will certainly make for a less noisy one. The amplitude level error of each sample is 256 times smaller at 16 bits and thus 1/256th as loud. While there is another divide by 256 going to 24 bits, the errors at 16 bit are already too small to be appreciable.

There can be some advantage in recording at 24 bit if one intends to do much processing. Mostly this is actually done in floating point (32 or 64 bit). With each DSP transform done on the data, new quantization errors are realized. Using a greater bit depth for processing keeps each error smaller, thus less noticeable.

As a practical matter, you could undoubtedly do half a dozen or more operations (noise reduction, declicking, EQ, normalization, etc.) on a 16 bit recording without being able to detect any difference over doing the same things after first converting to floating point. However, for a new project, starting by recording thirty, forty tracks or more, each of which may undergo a fair amount of processing, and which will eventually be mixed to a stereo master, the higher bit depth can be an advantage. Also, recording at 24 bit allows one to capture lower amplitude signals more cleanly, assuming one has a very quiet recording environment and the rest of the equipment is on a par.

The generally accepted process here is to make listening comparisons using ABX software. This assures you have no clues about which you are hearing except what you actually hear. You may find, as so many others have, that differences you believe you have heard when you knew what you were listening to are no longer there. Belief and expectation have powerful effects on our perception.

Analog Line in connection to e-mu 0404 USB?

Reply #17 –
Quote from: Cavaille on
Greynol is 95% right.

At 95% of "pretty much" you will get no argument from me.  I concede that some remasters are improvements, so long as they aren't squashed to death.  ADC has come a long way since the '80s, though some of the perceived improvement has as much to do with trying to avoid tapes that are generations removed from the original tapes as it does the equipment used to digitize it.  In the end, however, SBM releases (and the like) are still presented in redbook format.

Analog Line in connection to e-mu 0404 USB?

Reply #18 –
Quote from: greynol on
ADC has come a long way since the '80s,


If the target format is 16/44 and we're talking audible performance, then there has been zero improvement in the best available consumer ADCs and DACs since the 80s.  For example in 1973 I did digital recording and playback with an ADC/DAC that had true 16 bit performance and ran at 200 KHz. It also cost somewhere between a quarter and a half million dollars, was about 5RU had a big, fat power cord and lots of cooling fans. That's all that  improved about 16 bit equipment since then - the price, size and power requirements. There were some very suboptimal low cost 16 bit converters during the 80s and 90s, but they were justified by their relatively low cost.  At this point, even low-cost 16 bit converters deliver near theoretical performance.