Quadrature Input and RS422 Signal Levels

I have a quadrature signal coming from a motion control drive that is in RS422 levels. What would the easiest solution be to get those signals into the quadrature encoder module (cFP)?
Thanks,
Michael Wise

http://www.baldor.com/products/motioncontrol/microflex.asp?product.asp?1=1&product=Servo+Controls&fa...
It has a simulated (actual signal to drive is a resolver) quadrature signal coming out. I'm only concerned with A+B since I'm just using it to determine speed. In the spec it says the signal levels of the encoder output conform to RS422 Levels.
I was planning on using the NI cFP-QUAD-510 for reading the encoder A/B signals back into the cFP system.

Similar Messages

Recording quadrature encoder and AI signals simultaneously and generating output TTL signals in one VI

I want to record simultaneously encoder, pressure and other AI signals and output a TTL signal for every other pressure signal. I have two VIs: One is to record all signals and the other is to generate TTL signals.
1. Since I'm new to Labview, I would like to double check with experts that if my VI (attachment 1) did what I expect. I would like to have DAQ box read and write encoder signals and pressure signals at the same time so I can plot PV diagram.
2. During acquisition, some data points are usually missed, shown in attachment 2. Is there any way to solve it?
3. I tried to combine two VIs but it never work (attachment 3). Can anyone give me any suggestions?
PS. Labview 9.0 and NI USB-6216 are used.
Dummy signals are 30Hz sine waves for pressure and 2 TTL signals for encoder A and B at 5kHz.
Thank you!

Thank you for your help, Peter!
I attached the wrong file for attachment 1. The initial code for measurement is attached here.
The sampling rate is 20000 and the sample mode is continuous.
For missing data points, it improved when I add "wait until next ms multiple" in while loop.
Follow the example you gave me, I'm able to generate TTL signals in one vi, attachment 2. Could you please check if it's correct?
It looked fine with simulated signals and hope it will go well with real engine tests.
I have two more questions:
1. Does the input number for wait until next ms multiple should always be the same as "samples per channel" in sample clock?
2. How will the value of "samples per channel" in sample clock affect data acquisition? I tried 1000 and 10 but didn't see any difference between these two.
Thanks a lot.
Attachments:
recording VI.png ‏332 KB
AI+CI+CO.png ‏72 KB

M-Audio FastTrack USB - very low signal level (line/guitar)

I've used FastTrack USB for a couple of years with a microphone through the XLR input with no problems. Now I am trying to connect a line output of a synthesizer to the line input of the FastTrack and the signal level is very low. There is no noise, it's just quiet, even with the track volume all the way up it's still no match for the rest of the tracks.
Any suggestions, except for trying all the knobs and switches on the face and back of the unit (tried that already)? Anybody has similar setup?
I use FastTrack for audio input, and built-in audio for output, it's connected through a hub (no problems for two years). In GB microphone is going to the Channel 1 (mono), and still works fine. Synthesizer goes to Channel 2 (mono) and is really quiet. I tried the syn with two different 1/4 cables with no difference. The headphones jack on the front of the FastTrack works fine with the normal level adjustment range, and if I plug in the headphones directly into the line out of my syn, I have sound, quiet (like you would get by connecting headphones to any line out, turntable even) but steady, so the syn must be fine.
Any thoughts?

No takers. Hm. OK. After some more research on the Net it appears to be that all the Fast Tracks suffer from the same problem, which makes me believe that it's not a bug, but a feature of an unexplainable kind. Oh, well, back to my old iMic noisemaker (this one works, it's just screened very poorly, so it picks up all the possible interference, and also makes its own little nasty hiss--this is why I switched to the Fast Track a while ago--so I will need to cut it out with an EQ, again). Hopefully, the EQ surgery will not affect the synthesizer's sound consistency as severely as it use to do with a microphone.

Airport Signal Level

Hi all,
I have a wireless router in my office and things were going great until recently I notice a drop in speed. I checked my airport info and the signal level has dropped three bars. I don't recall making any changes to the router or to any settings.
My question is, what could trigger a drop in signal level? The router is right on my desk so I am pretty sure the distance is not the issue.
Any help on this issue would be greatly appreciated.
Respectfully,
Eric

They should be tunables on your wireless router. For Linksys routers, they are usually under the "advanced" settings of the wireless page.
xmit power is the power level in milliwatts output by your antennas so your airport card can receive transmissions.
the frag threshold is the max packet size used on the wireless network before it is fragmented and split apart, causing network drops and re-transmissions
the beacon is the emission from your wireless router to wireles network cards, sort of like: "who's out there", sometimes, more frequency can mean more reliable transmission, especially if you have neighbors.
You can find lots of info at dslreports.com in their various forums. I have had 0 issues with my settings, and I live in apartment with 15 other ESSIDs surrounding me.

Analog Levels vs SPDIF Levels Input and Output in Logic Pro

Hello,
I ran a test last night for recording input and output levels from my Yamaha Motif XS8 through an Apogee Ensemble to compare Analog to SPDIF
I connected two TRS cables from the L and R outputs on the Motif XS into the Analog Inputs on the Apogee and also have the SPDIF connection from the Motif to the Apogee.
I put the master fader all the way up on the XS for the volume for analog.
The ensemble in Maestro has a +4 and -10 reference notional level option for analog inputs. i had it at +4 but changed to -10 and the analog got louder (i figured it would get louder for +10, confusing).
anyways, why is it that I can record louder levels for analog than the digital transfer?
I tracked both options at the same time then recorded vocals over it. the digital sound is too low. what's up with that?
and when I tried to bounce the recording to listen to it in ITunes, the volume levels were way lower than my cd playing through iTunes. Please enligthen me on these?
I use to record in a Roland 2480, and had similar results with loudness, but got a little louder through mastering but still....pro cds are way louder and still clear.

I think there is much confusion here!
In summary, you wont be able to control the recording level of S/PDIF.
The reason is that you don't want to!
You need to think of the SPDIF connection as being more like a file transfer method. You are copying the digital data at an output to your harddrive in effect. If I send you an MP3 via email you'd never imagine that your email software is capable of changing the gain of the MP3 I send you. This might sound daft but its a useful analogy. If you need to increase the "volume" of that MP3 then you'd need to ask the sender. Its the same with your set-up.
There could well be somewhere on your synth that adjusts the instruments level, other than the master (analogue) output control. For example, make sure the midi volume of the intrument being played is set to full - ie midi vol 128. Perhaps there is somesort of virtual mixer onboard to control all the muti-timbral parts so make sure your part has its virtual fader turned up.
This is what (basically) is going on in the chain...
Your synth creates sound in its "digital brain". This sound is sent to an "output stage" which will distribute the sound to various outputs. In the case of the S/PDIF it will just send the raw digital data untouched. For the analogue side the digital signal (same as the one sent to S/PDIF) will be converted to analogue and then sent to a amplifier to get it to an appropriate "line level". This final level could well be controlled by an anogue volume control which could be adding more gain (than you think) too.
When things go to your sound card/ daw...
The purpose of a analogue gain control is to set the i/p signal so that it suitably loud to beat any noise that exists in your input circuits - so that a good signal to noise ratio is achieved. Analogue signals need to work in the right loudness zone (so to speak) as the analogue electronics will be designed to handle signal levels of a particular range. the gain control is there to make sure the signal is in that range.
Digital signals are far more predictable though and there is no advantage to your recordings if the incoming digital signal gets an increase of level at the input stage. All you are doing here is effectively adding a few zeros to the binary digital data!
Lets face it the point of recording is to get a copy of the original sound, that is as similar to the original as possible. With S/PDIF you get a perfect copy of what's coming out of your synth - so job's a good un!
If, when you come to mix in logic, you find the level of the digital recording is indeed too low for mixing/mastering purposes then just boost it in logic via a fader or via the gain plugin.
Those referrence values of -10dBV and +4dBu refer to analogue voltage levels only. they have nothing to do with the digital domain. The -10/+4 switch will be only relevant to analogue inputs and outputs. Using an analogue VU meter you should find that a sine wave that peaks at 0dBVU (totally different to 0dBFS BTW) is the equivant of a digital sine wave peaking at -18dBFS.
The analogue headroom (how loud you can go before things distort) depends on the analogue electronics and varies with different design. Analogue stuff, like mixers) often has headroom of 24dB or more. So that digital stuff can interface with analogue properly we allow for that analogue headroom to be around 18dB (usually enough in practise!)... hence -18dBFS(digital)=0dBVU(analogue).
To make your digital and analogue input signals sound similar in level you will probably have to reduce the gain of the analogue input. If you set the incoming analogue signal to peak around -14dB (or less!) or so you will probably find things more equal. If you are working in 24 bit your analogue levels can be seemingly very low before sound quality is affected. Its quite safe to record at -20 or even -30dB as shown on logic's meters for eg.
I hope all this waffle helps LOL!

How to get signal from analog input and send it to analog output (real-time)

Hi everyone,
I am doing simple task in Visual C++ and I am using PCI-6221(37 pin).
Basically, I want to send same signal from 'analog input' to 'analog output'
at the same time (almost), to make it real-time application.
Can someone please provide me sample program.
I would appreciate if you could provide me with the good tutorial which explains
step by step everything about programing NI-DAQmx for C/C++.
Best Regards,
Khassan
Solved!
Go to Solution.

This is my code in C++, you can optimize it if it looks too messy. This code reads signal from analog input and outputs it through analog output.
To make this code work additional include directories and library directories must be added from NI.
I hope it helps someone.
#include <stdio.h>
#include <conio.h>
#include "NIDAQmx.h"
#include <math.h>
#define DAQmxErrChk(functionCall) { if( DAQmxFailed(error=(functionCall)) ) { goto Error; } }
int main(int argc, char *argv[])
int32 error=0;
TaskHandle taskHandleRead=0,taskHandleWrite=0;
int32 read=0;
float64 dataRead[1000];
char errBuffRead[2048]={'\0'};
char errBuffWrite[2048]={'\0'};
bool32 done=0;
int32 written;
DAQmxErrChk (DAQmxCreateTask("",&taskHandleRead));
DAQmxErrChk (DAQmxCreateAIVoltageChan(taskHandleRead,"Dev1/ai0","",DAQmx_Val_Cfg_Default,-10.0,10.0,DAQmx_Val_Volts,NULL));
DAQmxErrChk (DAQmxCfgSampClkTiming(taskHandleRead,"",100.0,DAQmx_Val_Rising,DAQmx_Val_ContSamps,0));
DAQmxErrChk (DAQmxCreateTask("",&taskHandleWrite));
DAQmxErrChk (DAQmxCreateAOVoltageChan(taskHandleWrite,"Dev1/ao0","",-10.0,10.0,DAQmx_Val_Volts,NULL));
DAQmxErrChk (DAQmxCfgSampClkTiming(taskHandleWrite,"ai/SampleClock",100.0,DAQmx_Val_Rising,DAQmx_Val_ContSamps,1000));
DAQmxErrChk (DAQmxStartTask(taskHandleRead));
DAQmxErrChk (DAQmxStartTask(taskHandleWrite));
while( !done && !_kbhit() )
DAQmxErrChk (DAQmxReadAnalogF64(taskHandleRead,1,10,DAQmx_Val_GroupByScanNumber,dataRead,1000,&read,NULL));
DAQmxErrChk (DAQmxWriteAnalogF64(taskHandleWrite,read,0,10.0,DAQmx_Val_GroupByChannel,dataRead,&written,NULL));
_getch();
Error:
if( DAQmxFailed(error) )
DAQmxGetExtendedErrorInfo(errBuffRead,2048);
DAQmxGetExtendedErrorInfo(errBuffWrite,2048);
if( taskHandleRead!=0 )
DAQmxStopTask(taskHandleRead);
DAQmxClearTask(taskHandleRead);
if( taskHandleWrite!=0 )
DAQmxStopTask(taskHandleWrite);
DAQmxClearTask(taskHandleWrite);
if( DAQmxFailed(error) ){
printf("DAQmx Error: %s\n",errBuffRead);
printf("DAQmx Error: %s\n",errBuffWrite);
printf("End of program, press Enter key to quit\n");
getchar();
return 0;

Statistical forecasting....Input and Output at different levels? & MAD and MAPE calculations

Team,
Couple of questions on Statistical Forecasting...
1) Is it possible to have input and output KF at different planning levels?...
2) Is there anyway to see MAD and MAPE values for each Forecast run?
Any help is appreciated....Thanks for your help...
Thanks,
Krishna

Hi Krishna,
#1 - The Inputs and Outputs should be at the same base planning level. Output should be a Stored KF. The input can be a calculated KF.
To achieve input and output at different plan levels, you can do something like this:
I/P KF1@PL1 (stored) Output: KF3@PL2
Create a Calculated KF: KF2@PL2 with calculation like SUM(KF1@PL1) and use this as input for StatFcst
#2 - The result of Fcst run MAD cannot be seen from Excel Client yet, we will address this in near release. If you are on-premise, you can see the results in table: sap.sop.sopfnd.catalogue::SOPFCS_FORECAST_RUNTIME_INFO_TAB
Hope this is useful.
Thanks,
Raghav

Measure rpm and PWM signals using digital input of NI9401 module

Hi, is there a simple way how measure rpm (to 5000 rpm) and PWM (about f=3kHz) signals using digital input of NI9401 (ultrahigh-speed digital I/O) module plugged in NI 9172 ? I will wellcome any suggestions and VI examples (bether still). I'm begginer using LView 8.2. I apologize for my poor English. Many thanks.

Hi CUA,
Since you have already the right tools, you just need to get started with the software. The NI-9401 will give you access to the counters that are inside the cDAQ chassis 9172.
You need to place the module in slots 5 or 6. You can read in how they work and how to configure them in the counters section of the user guide found here.
Now to measure rpm, you can either measure the period of the signal (seconds) and multiply by 60 to have minutes and the number of ticks per revolution. You can find a shipped example to read period called “Meas Dig Freq-Buffered-Cont-Large Range 2 Ctr.vi” found under Hardware Input and Output » DAQmx » Counter Mesurements
To measure PWM, it is similar procedure and you can read the semiperiod (high time or low time). There is also a shipped example called “Meas Semi-Period.vi” found under the same location.
There is great information for these kind of task at this page or just post back if you need to.
Hope this helps.
Gerardo O.
RF Systems Engineering
National Instruments

TS1398 my new ipad works perfectly. the iphone 5, however, cannot connect to the same wifi, can barely get a 3G signal, and the power level is at 53% after being barely used for 1\2 a day. Is this a hardware problem?? My old 3Gs was fine until it died.

my new ipad works perfectly. the iphone 5, however, cannot connect to the same wifi, can barely get a 3G signal, and the power level is at 53% after being barely used for 1\2 a day. Is this a hardware problem?? My old 3Gs was fine until it died.

Greetings,
I've never seen this issue, and I handle many iPads, of all versions. WiFi issues are generally local to the WiFi router - they are not all of the same quality, range, immunity to interference, etc. You have distance, building construction, and the biggie - interference.
At home, I use Apple routers, and have no issues with any of my WiFi enabled devices, computers, mobile devices, etc - even the lowly PeeCees. I have locations where I have Juniper Networks, as well as Aruba, and a few Netgears - all of them work as they should.
The cheaper routers, Linksys, D-Link, Seimens home units, and many other no name devices have caused issues of various kinds, and even connectivity.
I have no idea what Starbucks uses, but I always have a good connection, and I go there nearly every morning and get some work done, as well as play.
You could try changing channels, 2.4 to 5 Gigs, changing locations of the router. I have had to do all of these at one time or another over the many years that I have been a Network Engineer.
Good Luck - Cheers,
M.

Required - Battery Level and network signal strent...

It would be very useful if there was a Battery Level and network signal strenth indicators in OVi Suite. IMO even free mobile management apps have this useful feature.So it shouldnt be a problem for Nokia to add this to Ovi suite.

When I connect my N8 to the PC I get the below window pop up.
So it seems the function you describe already exists.
Thanks
Ray

Audio input and output

I am trying to get audio to play through from the audio input to the audio output. I have plugged a line-level keyboard into the audio input and as I play it, it shows that it has plenty of input level in the sound panel of the system preferences. However, there is no sound coming out. I have tried speakers, headphones and the Griffin IMic. If I use a program like SoundStudio3 and click the audio "thru" button it works fine, but it does not work without.
Is there a THRU switch that I am missing somewhere? I tried the Applications/Utilities/AudioMIDI setup to configure it, but the Thru button at the bottom is greyed out (not in operation).
Thanks.
2 GHZ intel core duo Mac OS X (10.4.8)

Hi Rich,
Welcome to apple forum.
In my opinion, I don't think OS X have the capability to sent the audio input "THRU" to output directly (I could be wrong).
You need to use Garage band or third party software like you mention before to allow it.
Maybe if you connect third party device that is "supported" the signal become audible and you can hear it. Like mBox using core audio driver.
http://www.apple.com/macosx/features/coreaudio/
Good Luck.

Signal Level Meter

Signal Level Meter
CATV Signal Level Meter GAO2002 is specially designed for CATV system maintenance. Signal Level Meter GAO2002 features small size (160mm x 130mm x 65mm) , light weight (368g including battery), long operating time (more than 6 hours), and well-built from appearance to architecture. Signal Level Meter GAO2002 combines the most practical functions and makes them easy to use. What is even better about this piece of equipment is that its added aseismatic design makes it more durable than you can expect. Tests show that it can continue to work well after being dropped from 5 meters high. It supports Level, V/A, Tilt and Trunk Level testing as well as testing two channels at the same time.
CATV Signal Level Meter GAO2002 Specifications:
Frequency
Frequency range: 46MHz ~ 860MHz
Frequency step: 50kHz, 1MHz, 10MHz and 100MHz
Level Measurement
Range: 20dBuV ~ 50dBuV
Accuracy: ±20dB (20ºC ± 5ºC)
Resolution: 0.5dB
Carrier to Noise (C/N)
Option Range: 20dBuV ~ 50dBuV
Input signal range: ≥ 85dB
Measure accuracy: ± 2.5dB (20ºC ± 5ºC) shut off the carrier
Resolution: 0.5dB
Voltage
Input range: 1~100V (AC/DC)
Measured accuracy: ±2V
Resolution: 1V
Miscellaneous
Size: 160mm x 130mm x 65mm
Weight: 368g (including battery)
Working: -10ºC ~ 40ºC
Audio: Built-in speaker (Auto on in single frequency mode)
Battery
Built-in battery: 3.6V/2.1AH Ni-MH
Working time: More than 6 hours (Shut off the audio and LCD backlight)
Charging time: 10 ~ 12 hours (power off)
Gao Tek Inc.

Do you have drivers for LabVIEW?
~~~~~~~~~~~~~~~~~~~~~~~~~~
"It’s the questions that drive us.”
~~~~~~~~~~~~~~~~~~~~~~~~~~

Setup help - low signal level in Logic

I'm getting low input signal levels when recording in Logic (-17db to about -10db, on average), and am hoping someone here can point out an obvious oversight on my part.
I'm recording mic'd guitars and vocals (one track at a time), with the mic(s)running into a Mackie 1402 mixer via XLR. Signals are strong and fine when I solo on the Mackie.
The signal runs out of the Mackie via the Alt 3-4 bus into the first and second 1/4" inputs on the back of a Motu 828 mkII, then into Logic.
Even if I drive the signal to near clipping on the Mackie, I get these low levels when recording in Logic. This is true whether I'm using a condenser mic with phantom power for vocals, or a dynamic mic on a guitar amp. I've also tried swithcing XLR mic cables.
I'm running Logic Pro 7.2.3 and OS X (10.4.9) on a G5 2.5 DP.
Is my signal path in need of some tweaking, or perhaps it's a setting I'm overlooking on the Motu 828?
Any help greatly appreciated to get this rookie back on track!
G5 2.5 DP Mac OS X (10.4.9)

Hi,
You wrote:
I'm recording mic'd guitars and vocals (one track at
a time), with the mic(s)running into a Mackie 1402
mixer via XLR. Signals are strong and fine when I
solo on the Mackie.
The solo on the Mackie is probably louder,check to see if you have it AFL(after fader listen or PFL(pre fader level or "level set").
The signal runs out of the Mackie via the Alt 3-4 bus
into the first and second 1/4" inputs on the back of
a Motu 828 mkII, then into Logic.
The first thing to check is to see what kind of cables you are running into the Motu mkii.
A. Unbalanced (ie "guitar-type" with two wires)
B.Balanced (ie "stereo-type" with three wires)
The second thing is to check what your inputs on your Motu are,there are two choices: +4dBu("pro") or -10dBV("consumer")
Here's the catch:
You MUST MATCH the correct cables for the correct levels.If the Mackie is outputting -10dBV but your Motu is thinking it is +4 you see less level no matter how loud you make the Mackie outputs.
The Mackie can output AUTOMATICALLY either -10dBV or +4dBu depending on what cable is hooked up to it.
Even if I drive the signal to near clipping on the
Mackie, I get these low levels when recording in
Logic. This is true whether I'm using a condenser mic
with phantom power for vocals, or a dynamic mic on a
guitar amp. I've also tried swithcing XLR mic cables.
This is not where the problem lies.It lieas between the output of the Mackie and the input to Logic.
I'm running Logic Pro 7.2.3 and OS X (10.4.9) on a G5
2.5 DP.
Is my signal path in need of some tweaking, or
perhaps it's a setting I'm overlooking on the Motu
828?
Yes,a setting somewhere in the Motu is not set correctly.Check above for how to do that.
Any help greatly appreciated to get this rookie back
on track!
G5 2.5 DP
Mac OS X (10.4.9)
Cheers

Sound input and recording

Hello chaps,
I'm considering buying a Macbook at the weekend, and have been conducting a large amount of research. But I've not been able to find out too much about the sound on the unit. As sound recording would be a primary function of the unit, it would be most appreciated if somebody with a helpful bone or two could answer me a couple of questions!
Firstly, I understand the Macbook to have optical digital input and output.
What sample rates and bit-depths will the digital input lock to (and output at)?
Is it possible to have two separate audio applications recording from the optical input at the same time? What about the line input?
With regards to the line input, am I correct in assuming this is solely a line-level input, and a standard dynamic microphone cannot be connected (i.e. it must be an amplified microphone)?
And what about AirTunes? I quite fancy an Airport Express -- but understand this can only be used to stream music from iTumes. Is there really no way to have it receive and reproduce the actual "system" sound output?
Finally, I use Adobe Audition to do my recording under Windows. Without splashing out on ProTools LE or BIAS Peak, is there a suitable piece of software I might be able to obtain for less money? I am aware of Audacity, but was looking for something perhaps a little more professional with a good multi-track editor. (I'm aware this might be outside the scope of this forum area!)
If I do need to use Adobe Audition itself for some reason (perhaps dealing with a client's Audition Session file) I would be looking at dual-booting with Windows and transfering my licences for both Windows and AA to the Macbook; can anybody confirm that the audio I/O, both analogue and digital, works properly under Windows?
Many thanks for your time.

The digital noise is a factor of the number of bits. It is actually 6 * n, where n is the number of bits.
So for an 8 bit signal, the noise is 48 dB down, about par for a cassette tape.
16 bit is 96 dB down.
24 bit is 144 dB down, which for all practical purposes, means that everything else (pre-amps, et al) are what affects the signal fidelity.

12VDC for Limit and Home Signals for PCI 7334 using UMI-7764

Can 12VDC be used for Limit and Home Signals for PCI 7334 using UMI-7764.
Can the UMI 7764 be damaged if over voltage is supplied to it. If so can it be repaired or it has to replaced.

Specifications for the Limit Switchs (home, forward, reverse, etc) is
determined by the specifications of your NI motion controller PCI
board. The UMI functions only as a pass through and breakout for the
board .
The valid limit swich voltage for the PCI-7334 is 0 - 5 V. You will have to divide down a 12 V input before running it to
the UMI and to your board. The 3.3k pull up resistor is only used to
pull floating voltages to TTL levels and does not protect for high
voltage signals.
Please note that the UMI-7774 provides optical isolation for limit switches and extended input levels (e. g. limit switch voltages: On = 3.5 to 30 V, Off = 0 to 2 V).
Best regards,
Jochen Klier
National Instruments Germany

Quadrature Input and RS422 Signal Levels

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