# US-144 Calibration Question...



## etc6849 (Jan 4, 2009)

I just bought the US-144 and the ECM-8000 mic just to use the wonderful REW software on my xp based laptop  I'm a newbie, so please excuse any of my ignorance. I'm a pretty technical minded engineer though.

I had a few questions about calibrating the US-144 to determine if I need to order any additional cables:

1. I plan to use an XLR mic cable from the ECM8000 to the XLR input of the US-144. Will calibration through the left or right 1/8" analog input/output channels be good enough for the soundcard calibration? Obviously, there is some mic pre-amp circuitry that would be unaccounted for in the calibration right?

To account for the mic pre-amp circuitry in the US-144, should I use the soundcards analog output and feed this through the XLR input on the US-144 with phantom power off? Do you think this would increase the accuracy of the measurements or would it be a waste of money to buy a XLR to rca cable just for calibrating this way?

2. Does REW or others support response plots using an SPDIF output? If so, how would you calibrate this type of set up? For this I would use the SPDIF out on the US-144 or SPDIF on a soundcard combined with the XLR input on the US-144. 

3. What about running a test sweep through 6 analog channels of a ASUS Xonar card and using the US-144 as the mic input with one microphone at my main listening position? Is this possible or a good idea? I don't have a mixer or anything to provide phantom power is why I'm asking. 

I'm interested in this type of all channel test to tell what the response might be if I'm using all channel stereo on my receiver versus response for 2 channel stereo. I'm assuming it's easy for the software to accurately combine six individual response plots together, is this how a measurement like this is done?

Thanks in advance :bigsmile:


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## Wayne A. Pflughaupt (Apr 13, 2006)

etc6849 said:


> 1. I plan to use an XLR mic cable from the ECM8000 to the XLR input of the US-144. Will calibration through the left or right 1/8" analog input/output channels be good enough for the soundcard calibration? Obviously, there is some mic pre-amp circuitry that would be unaccounted for in the calibration right?


Yes. Our general recommendation has typically been to calibrate the regular line in/out and hope that the mic pre amp is flat. However, I personally don’t see why the calibration routine can’t be run from the mic in to the line out. In fact, it seems to me like it needs to be. Maybe John can shed some light.



> To account for the mic pre-amp circuitry in the US-144, should I use the soundcards analog output and feed this through the XLR input on the US-144 with phantom power off? Do you think this would increase the accuracy of the measurements or would it be a waste of money to buy a XLR to rca cable just for calibrating this way?


Yes, turn off the phantom power. As to whether or not it would be a waste of money for the cable, maybe you can be the first to give it a go!



> 2. Does REW or others support response plots using an SPDIF output? If so, how would you calibrate this type of set up? For this I would use the SPDIF out on the US-144 or SPDIF on a soundcard combined with the XLR input on the US-144.


I’ll defer that one to someone else more knowledgable.



> 3. I'm interested in this type of all channel test to tell what the response might be if I'm using all channel stereo on my receiver versus response for 2 channel stereo. I'm assuming it's easy for the software to accurately combine six individual response plots together, is this how a measurement like this is done?


Typically you want to measure one speaker at a time, with all processing turned off. But if you’re just looking to see what kind or response you’re getting with all channels running, just split the soundcard’s output to an L/R input, and then turn on the receiver’s all-channel stereo function.

Regards,
Wayne


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## etc6849 (Jan 4, 2009)

Thanks Wayne. I'm new to this stuff, but I'll give your suggestions to numbers 1 and 3 soon. I really like your idea for number 3. I'm also looking forward to doing tests with audyssey multiEQ on/off and see if audyssey really corrects for my room or not.

I'm still not sure if the US-144 limits the input frequency range for the mic inputs, but I know the FAQ's said something about never using a mic input on a laptop with REW. I'd think the XLR inputs on the tascam US-144 are much better mic input with fully range. I've also read a lot of people on avsforum are using this type of set up for portability. I couldn't find any specs for the US-144 mic XLR inputs other than these:

Audio specifications
Nominal input levels
MIC IN L and R (XLR balanced) –58 dBu (TRIM=max) to –14 dBu
(TRIM=min)
MIC/LINE-GUITAR L and R in
MIC/LINE position (1/4” jack, unbalanced)
–40 dBu (TRIM=max) to +4 dBu
(TRIM=min)
MIC/LINE-GUITAR R in GUITAR
position (1/4” jack, unbalanced)
–51 dBu (TRIM=max) to –7 dBu
(TRIM=min)
Maximum input levels
MIC L and R (XLR balanced) +2 dBu (TRIM=min)
MIC/LINE-GUITAR L and R in
MIC/LINE position (1/4” jack, unbalanced)
+20 dBu (TRIM=min)
MIC/LINE-GUITAR R in GUITAR
position (1/4” jack, unbalanced)
+9 dBu (TRIM=min)
Input impedance
MIC IN L and R (XLR balanced) 2.4 kΩ
MIC/LINE-GUITAR L and R in
MIC/LINE position (1/4” jack, unbalanced)
10 kΩ
MIC/LINE-GUITAR R in GUITAR
position (1/4” jack, unbalanced)
1 MΩ


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## JohnM (Apr 11, 2006)

Wayne A. Pflughaupt said:


> Our general recommendation has typically been to calibrate the regular line in/out and hope that the mic pre amp is flat. However, I personally don’t see why the calibration routine can’t be run from the mic in to the line out. In fact, it seems to me like it needs to be. Maybe John can shed some light.


There are 2 difficulties with calibrating a mic input. The first is level, there would need to be substantial attenuation between the line output and the mic input to prevent massive over-driving of the input. The second is the difference in impedances of the sources, i.e. the low impedance line output versus a high impedance mic, which may affect the frequency response. It is possible to make a cal measurement with an attenuator in front of the mic input (that's how I determined the grim response of the laptop mic input I tested) but it is probably only useful to see if the input has gross roll-offs at low and/or high frequencies.



> Does REW or others support response plots using an SPDIF output? If so, how would you calibrate this type of set up? For this I would use the SPDIF out on the US-144 or SPDIF on a soundcard combined with the XLR input on the US-144.


We don't generally recommend using an SPDIF output because of the calibration problem. One possibility is to loop back from an analog output of the receiver, but that can be complicated by the effects of crossover filtering - would need to ensure the output being used was configured for a full range speaker. Easiest is to just use an analog out from the soundcard.


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## Wayne A. Pflughaupt (Apr 13, 2006)

JohnM said:


> There are 2 difficulties with calibrating a mic input. The first is level, there would need to be substantial attenuation between the line output and the mic input to prevent massive over-driving of the input. The second is the difference in impedances of the sources, i.e. the low impedance line output versus a high impedance mic, which may affect the frequency response. It is possible to make a cal measurement with an attenuator in front of the mic input (that's how I determined the grim response of the laptop mic input I tested) but it is probably only useful to see if the input has gross roll-offs at low and/or high frequencies.


How 'bout if the soundcard (or USB interface, as I've seen some call them) had a balanced output? Could you do a mic pre-amp calibration then?

Regards,
Wayne


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## JohnM (Apr 11, 2006)

Still have the same level problem Wayne, the mic input will have anywhere between 20 and 40dB of gain so the line output would need a corresponding amount of attenuation.


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## etc6849 (Jan 4, 2009)

JohnM said:


> Still have the same level problem Wayne, the mic input will have anywhere between 20 and 40dB of gain so the line output would need a corresponding amount of attenuation.


Thanks John and Wayne for the interesting discussion.

Looking at the relevant specs of the US-144 I see:
Nominal input levels
MIC IN L and R (XLR balanced) –58 dBu (TRIM=max) to –14 dBu
(TRIM=min)
Maximum input levels
MIC L and R (XLR balanced) +2 dBu (TRIM=min)
Input impedance
MIC IN L and R (XLR balanced) 2.4 kΩ

and

Nominal output level
LINE OUT (RCA unbalanced) –10 dBV
Maximum output level
LINE OUT (RCA unbalanced) +6 dBV
Output impedance
LINE OUT (RCA unbalanced) 100 Ω

I think it's easy to mathematically show that I can/can't run a line from the line out to the mic input without clipping the mic input using the data above and the definitions of dBu and dBV (not the same as dBv). Once I figure out how to convert dBu to dBV I would know whether or not connecting them is a good idea or not. Using definitions found on wikipedia, it looks easy to do this. My thought is if the maximum dBV output from the US-144 is less than the mic maximum input (+2dBu), it's ok to do this, but may cause clipping. I'm guessing the best check would be to compare the nominal values for each. At any rate, once I do this calculation, I would know how much attenuation I would need.


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## brucek (Apr 11, 2006)

Well, it shows the maximum output line-out level to be ~ 2 voltsRMS (+6dBV), and the maximum input mic level to be ~ 1 voltRMS (+2dBu).

By those max values you would need to pad the input by 1/2, but I would go further than that (at least -25dB - you could require down to -40dB).

brucek


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## etc6849 (Jan 4, 2009)

Sorry I can't post the link (I don't think newbies are allowed to), but if you go on analog devices website they have a calculator that converts dBu to dBV. Converting the nominal output of -10dBV to dBu, I get: -7.782dBu. So since the nominal mic input with trim all the way down is -14dBu, I would need atleast -6.218dBu of attenuation (minimum). I would think you'd want to get in the middle of the -58dBu to -14dBu range anways, so add another 30dBu of attenuation and you get pretty close the attenuation John suggests.

Interesting though, -7.782dBu is within the max mic input of +2dBu, so I don't think it'll hurt anything to try it once, but what is the "nominal" volume level for the output? Where should my output volume be when I run these tests? any thoughts? My onkyo receiver says the rec out is [email protected] ohms, if I convert this to dBu I get -11.76 dBu so maybe this is the better way to go (assumes onkyo specs are in volts rms and a sinewave).



etc6849 said:


> Thanks John and Wayne for the interesting discussion.
> 
> Looking at the relevant specs of the US-144 I see:
> Nominal input levels
> ...


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## etc6849 (Jan 4, 2009)

brucek said:


> Well, it shows the maximum output line-out level to be ~ 2 voltsRMS (+6dBV), and the maximum input mic level to be ~ 1 voltRMS (+2dBu).
> 
> You would need to pad the input by 1/2, but I would go further than that.
> 
> brucek


My Onkyo 905 receiver says this for rec out: 
Output Level and
Impedance 200 mV/ 470
Ω
(REC OUT)

Following the same lines, is it ok to use this as the output to feed to the mic? I believe this would be a lot better choice over the soundcards output. .2vrms < 1 vrms, but .2vrms is not less than -14dBu~.1546 v rms and -14dBu is the nominal value.


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## brucek (Apr 11, 2006)

> My Onkyo 905 receiver says this for rec out:.....................Following the same lines, is it ok to use this as the output to feed to the mic?


You're not feeding the rec out to the mic input. The need for the pad is for when you do the one-time soundcard calibration with a short cable from line-out to mic-in of the soundcard. (personally, I think it's not needed and you can simply do a line-out to line-in cal and it would be fine, but that's another story).

As I indicated above, the max out of the line-out is a couple volts. That fairly standard. The nominal line-output is about 300mvolts (-10dBV).
A mic in is very sensitive. It shows a max input level of 1 volt (+2dBu), but that's quite high and would normally be within the ranges as shown from 150mvolts (-14dBu) down to 1mvolt (-58dBu).
So if you needed a pad to go from 300mvolts down to 1mvolt, it would require a ~-50dB down. If you wanted to go from 300mvolts down to 150mvolts, it would require a ~-6dB down.

So, as I calculated on the back of a napkin before, you probably would get away with about a -25dB pad. (I can tell you the resistors you would need)

brucek


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## Wayne A. Pflughaupt (Apr 13, 2006)

JohnM said:


> Still have the same level problem Wayne, the mic input will have anywhere between 20 and 40dB of gain so *the line output would need a corresponding amount of attenuation.*


Like this? (Balanced output)







​

Regards,
Wayne


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## brucek (Apr 11, 2006)

> Like this?


Twitchy is the first word that comes to mind..... 

brucek


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## etc6849 (Jan 4, 2009)

brucek said:


> You're not feeding the rec out to the mic input. The need for the pad is for when you do the one-time soundcard calibration with a short cable from line-out to mic-in of the soundcard. (personally, I think it's not needed and you can simply do a line-out to line-in cal and it would be fine, but that's another story).
> 
> As I indicated above, the max out of the line-out is a couple volts. That fairly standard. The nominal line-output is about 300mvolts (-10dBV).
> A mic in is very sensitive. It shows a max input level of 1 volt (+2dBu), but that's quite high and would normally be within the ranges as shown from 150mvolts (-14dBu) down to 1mvolt (-58dBu).
> ...


Thanks Bruce you have a great idea! What I was getting at is I could run the lineout from the soundcard into the receiver, then use the rec out to go into the mic input, then calibrate. However, I would still be over the max end of the nominal range for the mic input. 

I think I'll follow your suggestion of a voltage divider and aim for an input of about -22 dBu (about 61.5 mVrms) which is the middle of the mics nominal range (or should I be using 1mVrms??). As you state, I will assume a 300mVrms from the output. When I do this, I get these values: 1kOhm and 3.9kOhm (using standard resistor values). The 1kohm will go between the mic signal and ground. Any thoughts? What values were you thinking for the voltage divider? Thanks again.


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> The need for the pad is for when you do the one-time soundcard calibration with a short cable from line-out to mic-in of the soundcard. (personally, I think it's not needed and you can simply do a line-out to line-in cal and it would be fine, but that's another story).


I dunno, I just have a hard time believing and trusting that we can expect linear response from a budget product. Am I behind the times? I mean, I know that technology "trickles down," but has it trickled down all the way to the bottom end? :huh:

Regards,
Wayne


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> Twitchy is the first word that comes to mind.....


I have no idea what that means, at least as it relates to a level attenuator. Can you help me out? 

Regards,
Wayne


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## brucek (Apr 11, 2006)

> When I do this, I get these values: 1kOhm and 3.9kOhm (using standard resistor values). The 1kohm will go between the mic signal and ground. Any thoughts? What values were you thinking for the voltage divider?


Well, the 3.9K is fine (between the RCA output and mic input), but the 1K to ground is too high. This becomes your output impedance and would not work.
I suggest about a -32dB attenuation, which would be 3.9K (between the line-out and mic-in) and then a 100 ohm to ground (on the mic side of course). This would be about a -32dB pad (fairly typical for a mic pad), and would offer 100 ohm output impedance.



> What I was getting at is I could run the lineout from the soundcard into the receiver, then use the rec out to go into the mic input, then calibrate.


Yeah but then you've added the response of the receiver lein-in to your cal file, which you don't want to do......

brucek


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## brucek (Apr 11, 2006)

> I have no idea what that means, at least as it relates to a level attenuator. Can you help me out?


When you are trying to control a very high gain stage with a pot, the amount of change for even the slightest movement of the pot would be huge. You would likely be working with the pot almost in the fully CCW position, and the distance from too little to too much would be very small and would be considered _twitchy_.............. it would need a pad.

rbucek


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## brucek (Apr 11, 2006)

> I just have a hard time believing and trusting that we can expect linear response from a budget product. Am I behind the times?


No, I just think that in this case we would be compensating for the line-out stage, and the line-in stage, and would only miss the mic preamp (which feeds the line-in stage). Would it really make that much difference that we have to fabricate resistor pads for a dB or two ? When I look at the difference between my standalone soundcard cal file, and then the soundcard plus Xenyx802 mixer soundcard cal file, it isn't enough to even bother with. Do you think it would matter much if I added the mic stage into that file? I don't, and I don't bother..... I can move my microphone an inch or two when measuring and change the level by quite a few dB, so I don't think a dB at 10Hz being lost in my Mic preamp matters - the line stages are enough.

brucek


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## etc6849 (Jan 4, 2009)

brucek said:


> Well, the 3.9K is fine (between the RCA output and mic input), but the 1K to ground is too high. This becomes your output impedance and would not work.
> I suggest about a -32dB attenuation, which would be 3.9K (between the line-out and mic-in) and then a 100 ohm to ground (on the mic side of course). This would be about a -32dB pad (fairly typical for a mic pad), and would offer 100 ohm output impedance.
> 
> 
> ...


You are correct. This test jig's output impedance is very important to the mic's input. What is the typical impedance value for a omnidirectional condenser microphone? Specifically, I'm using the ECM8000. Mine hasn't arrived yet, but will be here tomorrow. I found this from google for the ECM8000:

Type: omni electret condenser 
Impedance: 600 ohms 
Sensitivity: -60dB 
Frequency response: 15Hz-20kHz 
Connector: gold-plated XLR 
Phantom power: 15-48V 
Weight: 4 oz. 

I suppose I should use a calculation like this to find the output impedance (of resistors and output of US-144) as seen by the mic's input:

100ohms (output impedance from US-144) + 3.9ohms = 4kohms. 4kohms in parallel with 100ohms gives 4kohms||100ohms ~ 97.5ohms. So really the mic input just sees 97.5ohms. 

Now, if I want to closely match the impedance of the ECM8000 I might try this (edited as I should have 4k = 3.9k + 100 (output imp from US-144): 
If my mic is 600 ohms, maybe a 750ohm resistor is ok to use as this would give (4*.68k)/(4+.68k)= .581kohms = 581ohms as seen looking from the mic's input into the test jig+output of the US-144.

Maybe I'm mis-understanding the mic's specifications... Am I missing something? Is the impedance of a condenser mic constant over it's bandwidth; ie should I be designing to this 600 ohms output impedance (test jig including output impedance of the US-144)?


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## brucek (Apr 11, 2006)

> 100ohms (output impedance from US-144) + 3.9ohms = 4kohms. 4kohms in parallel with 100ohms gives 4kohms||100ohms ~ 97.5ohms. So really the mic input just sees 97.5ohms.


Yes, that's correct. We're using this interface as a voltage bridge. It's a typical low output impedance feeding a high input impedance. The pad as I specified would honor that. 

Yeah, the mic is 600 ohms, but the impedance of the mic is not important enough to modify the pad to change the values as seen by the line-out in the soundcard cal routine. We are maintaining the 'rule of thumb' of >= 10:1 for a voltage bridge for both situations, and so it should work fine with the resistors specified.

brucek


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## etc6849 (Jan 4, 2009)

brucek said:


> Yes, that's correct. We're using this interface as a voltage bridge. It's a typical low output impedance feeding a high input impedance. The pad as I specified would honor that.
> 
> Yeah, the mic is 600 ohms, but the impedance of the mic is not important enough to modify the pad to change the values as seen by the line-out in the soundcard cal routine. We are maintaining the 'rule of thumb' of >= 10:1 for a voltage bridge for both situations, and so it should work fine with the resistors specified.
> 
> brucek


Bruce, what you are saying makes a lot of sense the more I think about it. Greatly appreciate your help and time. I'm going to use 3.9k and 100ohm resistors like you state. Thanks everyone for teaching me something. I must say hometheatershack is a pretty friendly forum.

Cheers,

Ellery


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> When you are trying to control a very high gain stage with a pot, the amount of change for even the slightest movement of the pot would be huge. You would likely be working with the pot almost in the fully CCW position, and the distance from too little to too much would be very small and would be considered _twitchy_.............. it would need a pad.


Makes sense. However, a mixer like the one I pictured is fully capable of accommodating a situation like that. In addition to the main output attenuator, you have post-preamp attenuators on the individual channel strips. Both are fully capable of reducing the incoming signal to zero. To address the "twitchy" situation, you can easily reduce the pre amp signal to a level that's extremely low via the channel strip fader (i.e., near full CCW). That leaves the main output fader with a wide sweep to fine tune the signal that's left.

Regards,
Wayne


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## etc6849 (Jan 4, 2009)

I'm kind of wondering the same thing. Guess I'll find out tomorrow when the US-144 and mic come in the mail. I'll post the cal results on here both ways (through input and also through resistor network and the mic) and hopefully one of you guys will take a look 

Cheers,

Ellery



Wayne A. Pflughaupt said:


> I dunno, I just have a hard time believing and trusting that we can expect linear response from a budget product. Am I behind the times? I mean, I know that technology "trickles down," but has it trickled down all the way to the bottom end? :huh:
> 
> Regards,
> Wayne


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## Wayne A. Pflughaupt (Apr 13, 2006)

brucek said:


> No, I just think that in this case we would be compensating for the line-out stage, and the line-in stage, and would only miss the mic preamp (which feeds the line-in stage). Would it really make that much difference that we have to fabricate resistor pads for a dB or two ? When I look at the difference between my standalone soundcard cal file, and then the soundcard plus Xenyx802 mixer soundcard cal file, it isn't enough to even bother with. Do you think it would matter much if I added the mic stage into that file? I don't, and I don't bother..... I can move my microphone an inch or two when measuring and change the level by quite a few dB, so I don't think a dB at 10Hz being lost in my Mic preamp matters - the line stages are enough.


No doubt, that's laudable performance on the part of the Xenyx 802. Maybe I'm just overly skittish, but I have my doubts that every other budget mixer or interface on the market is as good as Behringer's. There are an awful lot of them out there...

On the other hand, nothing in the signal chain here is laboratory grade to begin with, so maybe I'm being overly concerned... 

Regards,
Wayne


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## DrWho (Sep 27, 2006)

Hey etc....you're from the Klipsch forum right?

I think you might be way overthinking the issue....the US-144 has analog volume controls for the outputs and inputs. Just turn down the lineout.  The problem with an L-Pad is that it changes your effective output/input impedance in terms of where the LF corner ends up.

Also, when driving a balanced input with an unbalanced output, you will lose 6dB since you're tying the balance leg to ground with the adapater. I should have the opportunity to compare the line input to the XLR input, but I'd wager that the difference is trivial. It should be interesting to compare notes though.

The last set of measurements I did, I just used a loopback from the left output into the left unbalanced input. The ECM8000 was hooked up to the right channel xlr input. My measurements were very comparable to my M-Audio box that I had the opportunity to compare against the Klipsch anechoic chamber setup...that was within half a dB from 100Hz to 10kHz. Within a dB from 20Hz to 100Hz, and within 2dB above 10kHz.

The only source of frequency non-linearities in these devices is the digital anti-aliasing filters in the DACs/ADCs and then the series dc blocking caps. I suppose it's due process to ensure the XLR inputs have the same LF corner frequency as the line inputs, but I'd be surprised if it was very different. It would be pointless for them to not be using the same front-end for both the balanced and unbalanced inputs.

Ok, so I just looked at the spec sheet...unbalanced input impedance is 10k. Balanced input impedance is 2.4k. If the unbalanced input has a corner frequency of 3Hz, then the balanced input will have a corner at 13Hz. I just measured the corner of the line input at 3Hz...  I'll bust out an adaptor tomorrow evening and actually measure the LF corner of the balanced input. Anyone wanna take some bets?


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## brucek (Apr 11, 2006)

Wayne A. Pflughaupt said:


> In addition to the main output attenuator, you have post-preamp attenuators on the individual channel strips. Both are fully capable of reducing the incoming signal to zero


Look at the circuit again. The line-out of the soundcard will connect directly to the mic-in of the mixer, and the line-out of the mixer connects directly to the line-in of the soundcard. Those mains pots on the mixer are *after* the mic-in preamp and have no capability to dial down its input. The level into the mic mixer is set by the line-out sweep level adjusted in REW. You want to keep this high, and not dial it down to accommodate the next stage as a mic preamp or the level out of the soundcard would be too low. The S/N would suffer. So, the mic-in pot would stand by itself and would likely have to be dialed to almost zero to work (and would likely be twitchy). You really require a pad to dial that level down.



DrWho said:


> The problem with an L-Pad is that it changes your effective output/input impedance in terms of where the LF corner ends up.


Well, that's why I chose the values I did, to maintain the bridge impedance while supplying the required drop.



DrWho said:


> Also, when driving a balanced input with an unbalanced output, you will lose 6dB since you're tying the balance leg to ground with the adapater


But this is a non issue since REW normalizes the soundcard cal file to zero.



DrWho said:


> I should have the opportunity to compare the line input to the XLR input, but I'd wager that the difference is trivial.


Yeah, I agree, especially with this type of device. I think Waynes concern was that a cheap mic preamp may offer poor low frequency response.

brucek


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## brucek (Apr 11, 2006)

To prove that I eat my own cooking, I decided to test the pad I recommended on my own XENYX802 mixer.

I used whatever resistors I had in my drawer, and found a 100ohm and a 3.3Kohm - perfect, that calculates to -30dB pad with a decent output impedance (~100ohms) and an input impedance (~3.3K) that comes close to the mixers mic input stage (2.6K).

The XENYX mixer has a dual purpose pot that operates the input level for both the line-in stage and the mic-in stage, so I decided to leave the pot in its central detent position (the same as I always use).

Anyway, I did three measures. First the soundcard cal routine for a straight cable, then second for the mixer included into the line stage, then third, the mixer included for the mic stage. Before I ran the mic in test, I added a divider pad as described above and fed it to an XLR with pin 3 and 1 shorted to remove the negative differential amp and unbalance the input. I have to assume the response of the positive diff amp is identical to the negative amp.

It was interesting when I went to the mic-in test after the line-in test, I didn't touch the mixer controls to see how close the pad came to being right on the money, and it was very close. I only tweaked the mixer gain pot up a bit to bring the input level VU in REW to what it was before.

So, you can see the results below as expected. The mic stage response is down about -0.25dB at 10Hz. You'll note as expected that the mic response line (green) is noisier than the line-in response line (gold) or the soundcard only response line (purple). This could be cleared up if I didn't do such a quick and dirty job. My pad was simply clipped onto two RCA connectors and this is a no-no with a high gain stage such as the mic amp. You should put the pad in a metal box with short leads if you want a real clean line... (I don't really care myself - it's fine the way it is).

I show two scalings of the plot.

The first is with the standard +/-30dB we use for normal plots, so you can see the relative effect of the response curves.

The second is the expanded scale to get a close look..




















brucek


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## Wayne A. Pflughaupt (Apr 13, 2006)

> Look at the circuit again. The line-out of the soundcard will connect directly to the mic-in of the mixer, and the line-out of the mixer connects directly to the line-in of the soundcard.


Perhaps I was less than clear, the example I showed before is a USB interface mixer, not like the Xenyx. And unlike the Tascam US-144, it has balanced outputs. So the mic input would be fed from the _mixer’s balanced output_, not from an outboard sound card.










So, you have a fully balanced signal loop, which should eliminate the mismatched impedance issue John mentioned. And between the variable input pad and the two mains pots, that should address the level issue – right?

Not sure if this would address the noise issue you showed in your graph - you'd be a better judge of that than I.  As John noted, at least running a mic pre-amp calibration can show us if its response is reasonably linear, even if we can't get a usable calibration file... 

Regards,
Wayne


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## brucek (Apr 11, 2006)

> So the mic input would be fed from the mixer’s balanced output, not from an outboard sound card.


huh? we want to include the mixers response in the soundcard cal file. So instead of feeding the soundcard line-out to soundcard line-in (as per usual), we feed the soundcard line-out directly to the mic-in (not from the mixers line-out)...... the mixers line-out feeds the soundcard line-in.

brucek


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## Wayne A. Pflughaupt (Apr 13, 2006)

This mixer has a USB output for the computer. It _is_ the soundcard! IOW, it does what your Xenyx/soundcard set-up does, all in one unit instead of two...

Regards,
Wayne


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## brucek (Apr 11, 2006)

> This mixer has a USB output for the computer. It is the soundcard!


Oh OK, I see... huh, it looks just like the XENYX so I assumed it was the same....


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## Wayne A. Pflughaupt (Apr 13, 2006)

Soo... can an acceptable soundcard calibration be had between the balanced output and the mic input?

Regards,
Wayne


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## etc6849 (Jan 4, 2009)

Yep, that's me. Good to hear from you again. I think you were the one over there that told me about REW and I followed the links in your signature  

I am wondering about your statement of shifting the corner frequency. I think you have a good point.

Gain wise, you are just shifting the curve downward right by adding a padding circuit? This should not shift the corner frequency to the left or right since ideally I am adding only resistance.

My understanding is one should look at the gain from the output of the sound card, the gain from the voltage divider (padding circuit plus some cabling) and the gain from the soundcards pre amp. If all are in dB, you can add. If they are in voltage (vout/vin) you would multiply them. Since, with the voltage divider, I'm adding what should ideally be a constant impedance over all frequencies (no reactance, only resistance) I shouldn't see a corner shift left or right; only up or down. Now, if I open up the soundcard and modify the resistors connected to the op amps inside, yes this would shift the corner frequency. I wish I had a an adapter to connect this special padding circuit from the output to the input to verify this though cause you have an interesting point.

Bruce: Thanks for the results. I soldered the padding circuit up last night. I didn't use a metal box to shield the dividing circuit though, but just soldered and used shrink wrap to prevent ground to signal contact. I might use some foil and solder this to the ground later on. I don't know how that would work though as I've never tried soldering foil; I don't think it's worth finding a metal box for yet. My soundcard and mic are arriving today so I will post my results soon.



DrWho said:


> Hey etc....you're from the Klipsch forum right?
> 
> I think you might be way overthinking the issue....the US-144 has analog volume controls for the outputs and inputs. Just turn down the lineout.  The problem with an L-Pad is that it changes your effective output/input impedance in terms of where the LF corner ends up.
> 
> ...


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## brucek (Apr 11, 2006)

> Soo... can an acceptable soundcard calibration be had between the balanced output and the mic input?


Of course. You would use a partial H-Pad. The partial would be the best in this case. Simply take the values you would use for the simple divider (such as 3.3K and 100ohms for ~ -30dB) and divide the series resistor in half (i.e. 1650 ohms. Standard value of 1800 would be fine or go down to 1600).

The two 1800 ohm resistors would attach from the balanced line-out XLR pin 2 and 3 (or equivalent 1/4" phone), and then the other resistor of 100 ohms is soldered across the 1800 ohm resistor ends. Then feed the new balanced output from the 100 ohm resistor to the pin 2 and 3 of the mic-in XLR.

I could draw a picture, but I'm sure you get it......



> This should not shift the corner frequency to the left or right since ideally I am adding only resistance.


It won't, you'll be fine.



> I didn't use a metal box to shield the dividing circuit though


Not a problem, just keep the leads short and cover it with some foil or whatever. It'll be fine. I had 3 foot clip leads and it was sitting on my computer and mine worked fine....

brucek


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## etc6849 (Jan 4, 2009)

I've attached the measured file for one of my speakers. I had no idea how bad it would look. Is there something wrong with my setup? I plotted this after calibrating the ECM8000, adding loopback cable on the left output/input. I had the mic input on the card set on max, line out on max. I let REW do 4 iterations with 512k length.

I guess I really didn't realize there were 40 dB swings in my speaker's output. Is this normal or am I missing something? The green is without audyssey multiEQ, the blue is with Audyssey. The speakers are Klipsch RF-83's.

PS: Is there a best way to point the mic? Right now I'm pointing it straight at the speaker near the tweeters height, horizontal with the floor.


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## etc6849 (Jan 4, 2009)

Here's what I get for the soundcard with the padded loopback.


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## cinema mad (Jan 28, 2007)

the ECM 8000 mic should point up towards the ceiling at ear hight when seated, if you are checking the room response, it is designed to graze like the Audyssey mic....

Cheers..


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## JohnM (Apr 11, 2006)

The wild swings at higher frequencies are normal and caused by comb filtering, reflections from room surfaces partially cancelling the direct signal from the speaker at frequencies where the difference in arrival times corresponds to 180 degrees of phase shift. To see the underlying response apply some smoothing to the trace.

You will generally get perfectly good results with a single 256k sweep, multiple longer sweeps are useful in environments with higher background noise or if you need a very high signal-to-noise ratio for some post processing of the impulse response.


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## brucek (Apr 11, 2006)

> Is there something wrong with my setup?


No, it's your plot setup that needs to be changed.
For subwoofers, always use the standard Vertical graph axis of (45dB - 105dB) and the Horizontal graph axis of (15Hz - 200Hz) using the Graph Limits button in the top right corner of REW.
For full range, use the standard Vertical graph axis of (45dB - 105dB) and the Horizontal graph axis of (15Hz - upper limits you desire, i.e. 20KHZ - certainly no higher than your soundcard can extend).
For full range, enable smoothing to eliminate the comb filtering. Use a 1/3 octave smoothing.

brucek


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## DrWho (Sep 27, 2006)

etc6849 said:


> Is this normal or am I missing something?


This is absolutely normal.

I've gotta run, but you should read up on the Haas effect, Heyser, and Time Domain Spectrometry. Instead of smoothing your response, you should play around with the window on the impulse response.

What you are seeing in the frequency domain are the effects of the comb-filtering from reflections throughout your room. The way human perception works, we are able to filter out reflections and perceive the direct sound separately. There are of course limits to this which Heyser discusses to great extent. The reason you don't want to smooth the response is because the process doesn't differentiate between direct and reflected sounds. Sure, it'll give you a pretty line, but it will provide an inaccurate representation for what is actually being perceived. If you can control the influence of the reflections in the measurement, then you can get a better idea as to what your ears are actually hearing. The trade off, however, is a reduction in frequency resolution, so you can really only filter for the mid- and high-frequencies. This works out OK though because the Haas window is longer at lower frequencies.

I hope that helps at least a little bit - I could go on for hours on the topic, but I think it'd be best to study the original sources (as they are way smarter and usually offer better insight).

Btw, your soundcard calibration looks very similar to mine. I hope you're satisfied with the recommendation.


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## facesnorth (Aug 14, 2008)

I too recently bought a US-144 (and returned a TB-SRM, xenyx 802 & edirol usb-midi), mainly for these reasons:

1) it works with vista 64-bit (main reason I returned the TB-SRM)
2) I was taken by the all-in-one box which to me seemed more effective (and slightly cheaper) than all the cables, adapters, and seperate units I would have used otherwise.
3) thus it seemed with less equipment I would be getting higher quality, and a more elegant and portable solution.

I haven't had time to set it up or do anything yet. However, after reading this I'm worried it's going to be harder and less effective than I originally thought. I couldn't quite follow some of the posts about what actually was required to calibrate this thing. What needed to be soldered? I wonder if this is something I can do. I don't understand all the talk about padding and what to do with this thing once you make it. It sounds very unelegant which is exactly what I liked about this unit. Plus it sounds imprecise... Is there a way to be confident that I can get this thing properly calibrated and be able to trust my measurement graphs?

Would the EMU 0404 USB 2.0 have been a better choice? I can still return the US-144 if I need to. I'm also thinking of future uses down the road for this thing, and while I don't want to get carried away, I do have some interest in hobbyist sound recording as well.

I also have the ECM8000 mic and a RS analog SPL meter, a BFD, and all the necessary cables. I'm itching to get started already. One last question - I'm also using the Onkyo 905, which does the Audyssey MultiEQ. Should I do all my REW stuff before running MultiEQ and run MultiEQ after? Or should I run MultiEQ and then REW? Or should I not use MultiEQ at all?


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## brucek (Apr 11, 2006)

> I couldn't quite follow some of the posts about what actually was required to calibrate this thing. What needed to be soldered?.......................Is there a way to be confident that I can get this thing properly calibrated and be able to trust my measurement graphs?


The simplified lowdown on this issue is that there was concern that the response of the mic preamp integral to the soundcard, might affect measurements in REW. My feeling is that using the line-out to line-in when performing a soundcard cal test was suffice, and that the mic preamp would have no pronounced effect. I tested my XENXY to get a feeling how much of an effect it would have, and found it to be minuscule, with worst case being -0.25dB at 10Hz. This can be safely ignored.

We didn't get a reading from etc6849 on the differences he found in the US-144 between the line-in vs the mic-in, so I can't advice you on whether you need to include the mic preamp or not in the calibration. I suggest you do not.

The temporary mod requires constructing an attenuator out of wires and resistors to allow the inclusion of the mic amp into the calibrate test. I don't know if etc6849 used balanced or unbalanced. He may be able to tell you about that...

If you ignore the mic preamp in the test, you simply loop line-in to line-out temporarily to run the soundcard cal routine...

brucek


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## etc6849 (Jan 4, 2009)

Thanks everyone for all your help. I've learned a lot from everyone  It appears I have a lot of studying ahead to really understand all of the different plots in REW and how to properly use/interpret them. I've been busy lately with work, but I'll post a comparison plot of the US-144's calibration tonight or Saturday.

One thing about the US-144 is I am having problems using it on a dell laptop. Windows XP keeps telling me my usb device is drawing too much current (only when I plug it in), but the device works fine on my vista HTPC. The US-144 will work with my laptop, but sometimes I have to tell XP to reset the current protection on the usb port first. 

I'm thinking about returning it if I can't use it reliably with my laptop and getting a xenyx 502 and using it with my HTPC. Also, there is some noise at high volume levels when I use it with my laptop (possibly from the fan inside the laptop?). This all could be just my laptop going bad as it's 4-5 years old, but I don't really know.


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## facesnorth (Aug 14, 2008)

brucek said:


> The simplified lowdown on this issue is that there was concern that the response of the mic preamp integral to the soundcard, might affect measurements in REW. My feeling is that using the line-out to line-in when performing a soundcard cal test was suffice, and that the mic preamp would have no pronounced effect. I tested my XENXY to get a feeling how much of an effect it would have, and found it to be minuscule, with worst case being -0.25dB at 10Hz. This can be safely ignored.


Thanks, Bruce.


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## brucek (Apr 11, 2006)

> getting a xenyx 502


The 502 has no phantom power. You need the 802......


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## facesnorth (Aug 14, 2008)

etc6849 said:


> I'll post a comparison plot of the US-144's calibration tonight or Saturday.


Great I'd love to see this.



> One thing about the US-144 is I am having problems using it on a dell laptop. Windows XP keeps telling me my usb device is drawing too much current (only when I plug it in), but the device works fine on my vista HTPC. The US-144 will work with my laptop, but sometimes I have to tell XP to reset the current protection on the usb port first.
> 
> 
> > I was worried about this unit not have a power brick. I wonder if plugging the laptop in would help at all.
> ...


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## etc6849 (Jan 4, 2009)

I definitely do not fully understand the software, but here's what I was able to generate. I used the measure soundcard for calibration option under the settings menu. I had to increase the mic volume slightly for the padding (with resistors) option.

This was generated using a vista HTPC. The green is US-144 with normal loopback mono phono to mono RCA. The light blue is the ASUS HDAV1.3 card with normal feedback config. The dark blue is the US-144 with the mic preamp and padding cable included. I do see some difference, but I don't know how much it matters. The ASUS HDAV has a better response though I think, but it costs twice as much so go figure. I didn't change the default window, but then again I don't know if REW applies a window for soundcard cal measurements.

Hope this helps,

Ellery



facesnorth said:


> Great I'd love to see this.
> 
> 
> 
> ...


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## facesnorth (Aug 14, 2008)

etc6849 said:


> This was generated using a vista HTPC. The green is US-144 with normal loopback mono phono to mono RCA. The light blue is the ASUS HDAV1.3 card with normal feedback config. The dark blue is the US-144 with the mic preamp and padding cable included. I do see some difference, but I don't know how much it matters. The ASUS HDAV has a better response though I think, but it costs twice as much so go figure. I didn't change the default window, but then again I don't know if REW applies a window for soundcard cal measurements.


It's hard to tell what the colors are from the graph. Is the one up top that is closest to flat the ASUS?


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## etc6849 (Jan 4, 2009)

yes. it's kind of a blue green color. i guess i should've used different colors huh?


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## facesnorth (Aug 14, 2008)

I don't really know how this works or what makes a good device for use with REW, but that ASUS line sure looks good. Why did you get the US-144 if you have that ASUS? why not just add the xenyx802 for cheaper? Just so you could use your laptop and not have to move your desktop over by the a/v equipment? (that was my rational)

are you using vista 32 or 64? what other uses do you plan for the us-144? and why did you choose the us-144 over similar devices like the EMU 0404 or m-audio fastrack pro? thanks


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## etc6849 (Jan 4, 2009)

Well, my computer fans are a little noisy on the HTPC. I was hoping my laptop would work so I could have more portability and quieter operation. The US-144 is a fine card as the calibration will account for any deviations from 0dB that you see on my plots. You can always run a plot after calibration with a loopback cable to see; just clear the mic calibration file first though.

From Bruce's plots even if you have a very nice card (his looks even flatter than the HDAV1.3), the pre-amp will drop it down a little anyways. There are better and cheaper cards than the HDAV1.3 in terms of analog performance, I bought it because it's one of the only HDMI cards to bitstream TrueHD. If you look at some reviews, there are several cards with even better S/N ratios.

My laptop issues are probably my laptop. It's pretty beat up and the dvd drive doesn't even work. Personally, I don't think I would go with the ASUS line. I've had too many HDMI handshaking issues. If you are looking for a soundcard for an HTPC, wait a few more months until the new HDAV version comes out and drivers are not in beta.



facesnorth said:


> I don't really know how this works or what makes a good device for use with REW, but that ASUS line sure looks good. Why did you get the US-144 if you have that ASUS? why not just add the xenyx802 for cheaper? Just so you could use your laptop and not have to move your desktop over by the a/v equipment? (that was my rational)
> 
> are you using vista 32 or 64? what other uses do you plan for the us-144? and why did you choose the us-144 over similar devices like the EMU 0404 or m-audio fastrack pro? thanks


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## BigPines (Jul 10, 2007)

I didn't realize this discussion was going on and I posted regarding a similar issue with the M-Audio FastTrack Pro: http://www.hometheatershack.com/for...including-phantom-mic-input-3.html#post139665.

I am still trying to figure out if I have accounted for my mic pre-amp in my calibration procedure. I have been trying to understand everything that has been said here but I am at a disadvantage because I am not an electrical engineer. Here is the scoop:

The FastTrack Pro has two XLR and 1/4" TRS combo jacks as the only inputs. Page three of the manual says:

"Microphone/Instrument Inputs (Mic/Inst) – These Neutrik hybrid connectors will each accept a low-impedance mic level signal on *a standard three-pin balanced XLR or TRS plug*, or a high-impedance instrument level signal on *an unbalanced 1/4” TS plug*."

Page fourteen of the manual has the detailed specs:

Mic Inputs (A/D):
Input Impedance: 2.7k Ohms unbalanced, 5.4k Ohms balanced
Maximum Input Level: from +24dBu @ min gain, pad on to -40dBu @ max gain, no pad
Preamp Gain: >40dB
Pad: -20dB pad

Line Inputs (A/D):
Input Impedance: 20k Ohms balanced and unbalanced
Maximum Input Level: +4.1dBu balanced/+1.9dBV unbalanced
Pad -20dB pad

Line Outputs (D/A)
Output Impedance: 150 Ohms unbalanced, 300 Ohms balanced
Maximum Output Level: +1.8dBV, unbalanced/+10.1dBu, balanced

:scratch: So I am still confused. Given that I am going from the balanced 1/4" TRS outputs to the balanced 1/4" TRS inputs in the loopback without using the available pad attenuator or instrument switch, it would seem as though I am including the mic preamp but after reading this thread I just don't know.

Since it is the same physical input, I am confused about how I can know if I am using the input as a 1/4" balanced Mic input or a 1/4" balanced line input. I thought as long as I was not using the instrument level switch on the input, I was using the low-impedance mic level input.

Any help would be much appreciated.

Mike


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## DrWho (Sep 27, 2006)

Do you have any other USB devices plugged into your laptop? Every USB device needs to authenticate itself at a certain power level, which in the USB2.0 spec I believe is only one of two power levels. Your laptop's USB bus can only provide so much power, so each device is allocated only so much power. If you plug in an additional device asking for more than what isn't authenticated, then that device will not authenticate.

As far as the noise, that can only happen if your gain structure is incorrect (not sure what you're doing when you hear the noise). If the device has the 100 whatever dB dynamic range it claims, then it just means you're operating towards the bottom of the dynamic range of some device in the chain. You could also be imparting noise by forcing the laptop to attempt to provide power that it can't.

Btw, I measured -3dB at 3Hz with the unbalanced out to unbalanced in....am I reading correctly that you're showing -3dB at like 20Hz? Or is your measurement showing the XLR input?


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## DrWho (Sep 27, 2006)

facesnorth said:


> What needed to be soldered? I wonder if this is something I can do. I don't understand all the talk about padding and what to do with this thing once you make it. It sounds very unelegant which is exactly what I liked about this unit. Plus it sounds imprecise... Is there a way to be confident that I can get this thing properly calibrated and be able to trust my measurement graphs?


You do not need to pad or solder anything. I don't know why the discussion went down that path. Just attenuate the output so that you don't clip your input.

Technically you lose dynamic range when you do this, but the extra noise is in no way going to affect acoustic measurements in a normal listening environment. I hope my mentioning this doesn't turn this into an issue too - I'm just trying to be complete.


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## etc6849 (Jan 4, 2009)

Thanks Dr. Who,

You have found an issue with my US-144 set up in vista, but I don't know what's wrong. I really think I am using the software correctly, but yes the unbalanced line in/out seemed to fall near 20hz at -3dB as did the XLR padded loopback. This was on a vista pc. However, when I did the test on an XP dell laptop, -3dB fell around -4.5hz with the XLR padded loopback cable.

I'm quite confused now. REW said my levels were fine when I did the US-144 plot. The ASUS card has a response close to -3db @ 3hz so I think I am using REW correctly. Is there some set level I should have the US-144 set to when I do a calibration measurement? IE, should I have the line in @ 50% and the output at 100%?



DrWho said:


> Do you have any other USB devices plugged into your laptop? Every USB device needs to authenticate itself at a certain power level, which in the USB2.0 spec I believe is only one of two power levels. Your laptop's USB bus can only provide so much power, so each device is allocated only so much power. If you plug in an additional device asking for more than what isn't authenticated, then that device will not authenticate.
> 
> As far as the noise, that can only happen if your gain structure is incorrect (not sure what you're doing when you hear the noise). If the device has the 100 whatever dB dynamic range it claims, then it just means you're operating towards the bottom of the dynamic range of some device in the chain. You could also be imparting noise by forcing the laptop to attempt to provide power that it can't.
> 
> Btw, I measured -3dB at 3Hz with the unbalanced out to unbalanced in....am I reading correctly that you're showing -3dB at like 20Hz? Or is your measurement showing the XLR input?


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## etc6849 (Jan 4, 2009)

I'm by no means an audio person, so I don't know how much I can help. 

I can say that I have also tried an m-audio buddy (little preamp) with the my vista computers soundcard. For the audio buddy I decided to try an XLR to rca cable without padding to see what would happen and I put this cable between mic input on the preamp and the soundcards right channel output. I also put a cable from the output of the preamp to the input of the soundcard.

I had to set the computers line in to 15% and the mic level to about 20% to keep the necessary headroom when I used a non-padded cable (ie cable without resistors). The calibration data was close to when I used the padded cable Bruce helped me with, so I'm not sure if you need to do it or not. I had to adjust level a lot lower to avoid using the non-padded cable and almost couldn't do it, even with REW output set to -20dB. 

In contrast with the padded cable, I set the computer line in to 100%, REW soundcard calibration to -14dB, and the m-audio buddy to 60% mic level. The only thing I noticed about the pre-amp versus the ASUS HDAV1.3 soundcard with a standard loopback going from in to out is that there is -15dB difference at very low frequencies under 10Hz, so I'm not sure it matters unless you are trying to measure frequencies under 15Hz.



BigPines said:


> I didn't realize this discussion was going on and I posted regarding a similar issue with the M-Audio FastTrack Pro: http://www.hometheatershack.com/for...including-phantom-mic-input-3.html#post139665.
> 
> I am still trying to figure out if I have accounted for my mic pre-amp in my calibration procedure. I have been trying to understand everything that has been said here but I am at a disadvantage because I am not an electrical engineer. Here is the scoop:
> 
> ...


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## DrWho (Sep 27, 2006)

etc6849 said:


> Thanks Dr. Who,
> 
> You have found an issue with my US-144 set up in vista, but I don't know what's wrong. I really think I am using the software correctly, but yes the unbalanced line in/out seemed to fall near 20hz at -3dB as did the XLR padded loopback. This was on a vista pc. However, when I did the test on an XP dell laptop, -3dB fell around -4.5hz with the XLR padded loopback cable.
> 
> I'm quite confused now. REW said my levels were fine when I did the US-144 plot. The ASUS card has a response close to -3db @ 3hz so I think I am using REW correctly. Is there some set level I should have the US-144 set to when I do a calibration measurement? IE, should I have the line in @ 50% and the output at 100%?


That almost sounds like a latency issue maybe? Did you have REW set to a sample rate of 44.1kHz? I don't think the Tascam does 48kHz (but that wouldn't explain a 2 octave shift).

If I'm remembering correctly, I run the line out at 12-o'clock and both line-ins at 3-o'clock. But really, the actual volume doesn't matter so much as long as you're above the noise floor. I run it at the settings I do because I get a full volume from the microphone at 3-o'clock as well. In other words, I don't think that is the problem.

I didn't get the chance to measure the XLR input last week, but I'm hoping to get around to it sometime this week. Maybe I will detail the process and throw it up on my website - dunno if that will clear things up or not.


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## brucek (Apr 11, 2006)

> Is there some set level I should have the US-144 set to when I do a calibration measurement?


Take at look at the REW scope results after the measurement and be sure it looks OK.. (i.e. proper input output levels, no clipping, etc)

bruce


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## etc6849 (Jan 4, 2009)

brucek said:


> Take at look at the REW scope results after the measurement and be sure it looks OK.. (i.e. proper input output levels, no clipping, etc)
> 
> bruce


What a practical way to check output/input levels. I guess I need to become more familar with the software because I didn't even think about looking at this!

Thanks,

Ellery


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## etc6849 (Jan 4, 2009)

Thanks Dr. Who. I've been busy these last few days, but I'll check to see what I have the sampling frequency set to. I remember the US-144 being set to 44.1khz in vista, is 48khz better? What about using 96khz or 192khz (the asus card will do this)?

If I understand the theory correctly, a sample rate of 44.1khz should be fine for frequencies upto 21.55khz right (nyquist theorum, about all I remember from a linear systems)? Are you saying that 44.1khz is not as good as 48khz if the DACs sample at 96khz? I'm pretty confused on this issue. However, I always wondered why receivers have 192khz DACs though when DVD's and most blu-rays are only sampled at 48khz/channel. I'll have to read up on this when I find some time... I'm assuming the added bandwidth leads to a better S/N ratio?

Cheers,

Ellery



DrWho said:


> That almost sounds like a latency issue maybe? Did you have REW set to a sample rate of 44.1kHz? I don't think the Tascam does 48kHz (but that wouldn't explain a 2 octave shift).
> 
> If I'm remembering correctly, I run the line out at 12-o'clock and both line-ins at 3-o'clock. But really, the actual volume doesn't matter so much as long as you're above the noise floor. I run it at the settings I do because I get a full volume from the microphone at 3-o'clock as well. In other words, I don't think that is the problem.
> 
> I didn't get the chance to measure the XLR input last week, but I'm hoping to get around to it sometime this week. Maybe I will detail the process and throw it up on my website - dunno if that will clear things up or not.


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## DrWho (Sep 27, 2006)

I mentioned the 44.1kHz issue because I was getting weird results with the 48kHz setting, but that was on the very first day I plugged it in and I was still messing with the driver. I should see if I can get 48kHz to work.

The only advantage to higher sampling rates is that the lowpass filters become easier to implement. You don't usually see a significant reduction in the noise floor because most ADCs and DACs are way oversampling at rates like 300kHz+ internally. You really don't want to go above 96kHz for the data though because you lose bit accuracy and it's harder to maintain clock accuracy as you go higher in frequency (since your PLL loops have gotta have way higher bandwidths to keep the jitter low).

As a measurement tool, 44.1kHz is totally sufficient as the passband ripple is less than 0.1dB (or very close). I doubt very much that the microphone is that accurate, but even if it were, there are no speakers that will be anywhere close....and it's not like the big problems that need fixing are going to be addressed to that accuracy either.

I've been too busy to even pull my measurement rig out of my car, but eventually I'll get around to testing everything you've been doing so we can compare notes.


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## mswlogo (May 8, 2007)

I use the TASCAM US-144 with the ECM-8000 Mic and it works fine.

You have to fiddle with levels a little so that you get good dynamic range on the inputs to the US-144 (i.e. just under the lights going Red Yellow) and getting REW happy with the levels. And not driving the speakers to low or to hard. You have to set input sensitivity different for the Mic vs the feedback.

I have used it with the Left Channel feedback loop for sound card calibration.

Also when you do the feedback you use the 1/4" plug and I believe that will connect to a line level input. When you connect the Mic you use a Balanced Cable which will connect to the Mic Preamp. You also need to turn on Phantom Power.

You won't be able to calibrate out the difference between the Mic input and Line Level input but I don't think that's a huge deal.

I too had to fiddle with 44.1Khz vs 48Khz and I forget which worked better.

Because REW can't support ASIO (Yet ) you are still going through the computers Mixer which often resamples to 48Khz. I think this is the reason why 48 vs 44 makes things work well or not. Don't forget the Levels set it the PC's Mixer are also in effect.


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## mswlogo (May 8, 2007)

facesnorth said:


> are you using vista 32 or 64? what other uses do you plan for the us-144? and why did you choose the us-144 over similar devices like the EMU 0404 or m-audio fastrack pro? thanks


I own both the EMU 0404 and the Tascam US-144. I like the TASCAM US-144 better. Under Vista-32 EMU 0404 is buggy. Under Vista-32 Tascam also has issues with 1.12/1.122 Driver/Firmware but 1.11/1.11 works good. The 1.12/1.12 Firmware/Driver was released to support Vista-64 and don't know how stable it is under 64bit OS. When I say not stable with 1.12/1.12 on US-122 it will hang playing music once every few hours. EMU hangs once every few songs. Both are easy to recover and just annoying.

I use TASCAM as Analog Phantom Powered Mic Preamp for Merdian Room Calibration with ECM-8000.
I use it for REW.
I use it as an SPDIF output into Meridian Digital Speakers (BitPerfect using ASIO) using Winamp.
I use it as a HeadPhone amp using Winamp (again with ASIO and plays 24/88.2 files just fine with no resampling).
I use it to record 24/88.2 SACD out of a PlayStation3 with a Modified CLUX11SA (taps SPDIF off HDMI).
I use it with an FM Tuner to feed Meridian SPDIF input only speakers.

I really like the TASCAM being USB powered.

I also bought the Creative Audigy and the M-Audio units and returned both. Real junk in my opinion. Might be fine for REW but not for a lot of other duties.


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## DrWho (Sep 27, 2006)

Oh cool, I didn't even realize the 144 could do ASIO. Where did you get your winamp driver?
_Posted via Mobile Device_


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## mswlogo (May 8, 2007)

DrWho said:


> Oh cool, I didn't even realize the 144 could do ASIO. Where did you get your winamp driver?
> _Posted via Mobile Device_


From here http://otachan.com/out_asio(exe).html

It inlcuded source and I downloaded the ASIO SDK and rebuilt it and fixed a few small bugs. Turns out most of my problems were the latest Vista driver from TASCAM. The 1.11 driver firmware is much more stable.

I'll put it up the cleaned up ASIO winamp plugin on my own site when I get some time. The main bug I fixed was it would crash when switching sample rates (in gapless mode).

http://www.meridianunplugged.com/wiki/index.php?pagename=Using a Computer Direct to DSP Speakers


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## The Bogg (Mar 20, 2009)

I just got the US-144 to use on a laptop with Win XP SP3. When I try to calibrate the soundcard it doesn't give anything close to a flat frequency response. Above about 600hz it starts to oscillate and is off by 3db or more at higher frequencies. I'm only using it for 400hz and under but still wonder why this is the case. I'm probably doing something wrong but I've followed the instructions in the help file to a T.
I upgraded to the latest 1.12 driver and firmware. Anyone have any suggestions? I tried initially doing the loopback with a cable with rca connector on one end and xlr on the other and I used the mic input. Since the result was off I suspected it was my use of a hybrid cable. I got the rca to trs adaptor and use the regular line in but the same result...ie. not even close to flat from 20hz to 20khz. Any suggestions would be appreciated!


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## brucek (Apr 11, 2006)

> I tried initially doing the loopback with a cable with rca connector on one end and xlr on the other and I used the mic input. Since the result was off I suspected it was my use of a hybrid cable. I got the rca to trs adaptor and use the regular line in but the same result...ie. not even close to flat from 20hz to 20khz. Any suggestions would be appreciated!


The cable should be an RCA to TS adapter, not TRS.

brucek


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## The Bogg (Mar 20, 2009)

brucek said:


> The cable should be an RCA to TS adapter, not TRS.
> 
> brucek


I meant it was a phono to 1/4" mono jack - the same one that was listed in the help file.


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## brucek (Apr 11, 2006)

You do need to connect from line-out to line-in on a single channel only and be in stereo mode with any effects or mixers in the application software turned off.

If you've done all that, then it should work. Try adjusting the input level down a bit when setting up the soundcard calibration routine to see if that helps the oscillation.

brucek


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## The Bogg (Mar 20, 2009)

brucek said:


> You do need to connect from line-out to line-in on a single channel only and be in stereo mode with any effects or mixers in the application software turned off.
> 
> If you've done all that, then it should work. Try adjusting the input level down a bit when setting up the soundcard calibration routine to see if that helps the oscillation.
> 
> brucek


Thanks for your reply. I do have only the left line-out to line-in. I'll recheck the other settings.
In any case, it seems linear enough from a few hz to 500hz which is really what I'm interested in
but it is a bit puzzling...


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