SIngle reference for PXI 6509 Input and Output

Hi ,
I am using 96 channel PXI 6509 as a DIO, In this 96 channels i need to assign 1st port for input, 2nd port for output and 4th and 5th for input so on. its is not a big deal using labview but labview creates seperate reference( Task in) for input and ouput.
I need to have a single reference for both input and output ( like we have in NI DCpower for SMU)
Can any one suggest me how i can achive this?
Thanks and best regards,

I would accomplish this with an Action Engine that handles all of your tasks. Alternatively, make a class that handles all of the tasks.
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines

Similar Messages

SNMP OID for 5108 Chassis input and output Watts?

Anybody know these?
Craig

Craig,
OID for cucsEquipmentChassisStatsTable -- 1.3.6.1.4.1.9.9.719.1.15.10
UCS MIBs are available here
ftp://ftp.cisco.com/pub/mibs/supportlists/ucs/ucs-manager-supportlist.html
HTH
Padma

Input and output on same device, producer/consumer structure

Hello interested people,
I have a question about using the same device for both digital inputs
and outputs. I have written a simple program of one while loop
that continuously polls the device, processes, and requests. I
have addressed the device using two DAQmx Asst. and I have attached
them with their error in/out cluster terminals to provide data flow and
eliminate the chance of addressing the devices at the same time (which
produces an error). Now I want to change this program structure
to a producer/consumer loop foundation with state machine.
In this design, I will have the DI in the producer loop and the DO in
the consumer loop, under one of the states. I can't simply
connect the error in/out ports in this configuration, so my question is
how to avoid the error caused by addressing the same device
simultaneously with two different tasks (input and output)? I
have attached two VI's, the "One Loop" vi is the original configuration
(simplified), and the Producer-Consumer vi is a NONSENSICAL program
that simply represents the desired configuration. (I don't need
any comments on the programming of this vi, it is only an example for
illustration of the problem).
I am thinking about bundling the input data and the error cluster, both
from the PXI 6528 DI, into one cluster, queueing that up, and
unbundling the de-queued elements for some kind of data flow between
the producer loop and the "Request" state of the consumer loop.
Is this the right approach, or am I barking up the wrong tree?
Thanks
Attachments:
One Loop DO DI.vi ‏102 KB
Producer-Consumer DI-DO.vi ‏106 KB

Hello,
It sounds to me like you really have two modes:
1. user interface actions determine execution
2. user interface is locked, and execution is automated.
I think it would make sense to use the producer consumer for an architecture. Basically you would do the following:
1. program the producer to handle the user interface as you normally would.
2. provide one additional event case in the producer which would be your "automated handling" case. In that case, you could put a state machine which could run until whatever conditions were met to put your program back in "user interface mode".
Keep in mind that you can use custom USER EVENTS to programmatically generate events ie. you can trigger the start of your "automated handling" form anywhere in your code at virtually any time.
I think this would allow you to take advantage of the producer consumer architecture in its intended spirit, while integrating an automated routine.
I hope this helps!
Best Regards,
JLS
Best,
JLS
Sixclear

Analog Input and Output in One Single VI

I need help in setting both analog input and output in one single VI. How do I assign channels to be either input or output? How do I simultaneously uses both in one single VI with a while loop structure?? Which AO am I suppose to use to obtain signal from the function generator I have built to feed into the DAQCard-1200?? Help!!
Attachments:
Test1.vi ‏48 KB

One thing you'll need to be aware of is that you will need to DMA lines: one for AI and one for AO. If you don't, then you can configure the DAQCard to do without DMA using the Config VI. But you certainly can do this.
As far as your function generator, you will want to do a buffered analog output. You will write your buffer of points to the buffer, and then tell NI-DAQ how fast to update your analog output channel with these values.
So, you can be reading from AI and checking the AO process in the same while loop. Just make sure you handle the while loop execution (the wait it exits) correctly. This can get tricky when you're doing two types of measurements.
J.R. Allen

How can I use a different driver for audio input and output?

I did a search of course, and came up with something about an aggregate. I have no idea what this is, how to do it, or if it would even work for me.
What I am trying to do is:
1) Record into Logic Express using my Tascam US-122.
2) Have playback come out of my computer sound system, not the Tascam.
If I go over to the Audio setup window, I can only record when the driver is set to Tascam US-122. Likewise, I can only listen to sound when my Built-In Audio is selected. It gets rather annoying going between the two.
So, would this aggregate thing solve my problem? If so, how do I do it? Thanks for any help!
-allen

Yes it should do what you want.
Go to "Audio Midi Setup", and go to the Audio menu and click "Open Aggregate Device Editor". The interface is pretty simple but if you do get stuck, just use the help function in Audio Midi Setup, as it has a step by step guide.
Then when you return to logic, go to the Preferences>Audio>Drivers section and select Aggregate Device as the new driver rather than either the built in sound or the tascam. Then the inputs and outputs will apply to BOTH devices.

How can I make Apple Earphones be the Input and Output for Audio

I have some older headphones and they aren't compatible with a Mid 2010 (Intel Core i3) 21.5 Inch iMac
Is there some way to make Apple Earphones be the Input and Output for Audio in the iMac
This is a problem for me personally as when I am talking to someone on Skype the iMac picks up the In-built Microphone audio, and yes the headphones I'm using do work straight from the iMac as audio output
Any help is appreciated

you can connect the headset to something like this
http://www.ebay.com/itm/NEW-MaelineA-3-5mm-Female-to-2-Male-Gold-Plated-Headphon e-Mic-Audio-Y-Splitter-/381100440348
and then connect the mic to the input and the headset to the headset port

How to use a bluetooth cellphone for audio input and output

Hello, I was wondering how to set-up my Mac Book so that I can use my bluetooth cellphone for audio in and audio out. Ideally, I would like to use my handset for talking on iChat and Skype. Has anyone had any luck doing this? When I bond my cellphone these optiosn do not appear. I know that this is easy to do on a Windows machine but I am not sure about how to do it on my Mac Book.
Thanks!

Yes it should do what you want.
Go to "Audio Midi Setup", and go to the Audio menu and click "Open Aggregate Device Editor". The interface is pretty simple but if you do get stuck, just use the help function in Audio Midi Setup, as it has a step by step guide.
Then when you return to logic, go to the Preferences>Audio>Drivers section and select Aggregate Device as the new driver rather than either the built in sound or the tascam. Then the inputs and outputs will apply to BOTH devices.

I have on;y one audio port can i use it for input and output at the same time?

i have the new macbook pro 13 in and i want to connect aguitar as input and output on an amp but i have only one audio port

You can't use the jack as an input and output simultaneously. The solution is to look for a USB or Firewire interface to get your guitar sound into the computer.

(Input and) output formats from Photoshop Elements 6 for Mac

I noticed that (until midnight PT, anyway) Amazon has Photoshop Elements for the Mac at a lower price than usual. This brings up a question for me: What output formats can Photoshop Elements 6 produce? In particular, can it produce PNG? Also, can it accept PNG as input?

I think that all the formats like .bmp, camera RAW, EPS,PICT, TIFF, Filmstripn, IFF, JPEG, Photoshop pdf, Photoshop RAW,PNG etc are supported for both input and output.

When you set up tracks what is best to pick for the input and output?

When you set up tracks what is best to pick for the input and output? also what is the buss output? Thanks

First part of the question I don't understand, so could you elaborate please?
Second part: bus output is used for submixes, for example, let's say you want the same EQ to be applied to all guitars in the song, so instead of using same EQ plug-in on all guitar tracks, you feed the output from these tracks to a bus track and use the effect just once on this bus submix.

I'm trying to record in Adobe Audition and it keeps saying my the sample rates for my input and output devices do not match. How do I correct this?

I'm trying to record in Audition and it keeps saying my sample rates for the input and output devices don't match. How do I fix this?

I finally have communication between the ISA One and Audition. I moved the optic cable to another SPDIF on the computer and from the SPDIF to ADAT on the ISA One and it finally agreed that the communication was at the same clock speed. The audio was not intelligible so I moved the optic cable back to the original configuration and I can record voice that is clean.

CFP-AIO-610 electrical isolation between input and output channels

Manual states "The input and output channels of the [c]FP-AIO-610 are not electrically isolated from each other". I assume this means that the commons on the input channels are connected and the commons for the output channels are connected but it does not mean that all channels on the module are connected. Is this correct? I intend to monitor two self-powered 4-20mA signals coming from a single instrument and also send two voltage signals to a different instrument. Any advice is greatly appreciated. Thanks!

RavensFan wrote:
Why do you think there aren't situations where common AI and AO references are acceptable? I think in most situations where it is already okay to use non-isolated channels, having common commons is just fine.
From the standpoint of interacting with two different instruments, such as my situation. Rarely in my experiences so far are analog communication or measurements on separate instruments going to all be referenced to the same common / earth ground. Instrument A has it's own common reference and instrument B has it's own common reference. My 20mA loop is externally powered and I can earth ground the power supply negative but the voltage output to another instrument is not but hte channels are isolated so maybe I can do it anyway. I'm hardly an expert and only trying to build my own knowledge base on how to use NI hardware. Perhaps I'm over-thinking this too much and it will work.

How to temporally match/save Input and Output Channels's data?

Hello, I have a voltage-input, voltage-output SISO system, and need to indentify its dynamic response (or transfer function) as a reference to a PID control. Without using the system identification toolbox, my goal is to generate/provide a sine sweep input (ranging 1Hz-5kHz for 10sec) to the system and to save the corresponding output response signal as well as the sweep input signal simultaneously.
I got a sample program online and am trying to modify it as attached, but I really need your advices/comments concerning several problems I am facing with:
1) With the below setting, the number of the acquired Input channel samples obtained is 5461, which is smaller than expected (i.e., 10000). What would I be missing in setting parameters?
- H/W: NI-USB-6341 DAQ
- AO: continuous samples with the waveform timing (1k sampling, 10000 samples => 10 sec, slower sampling just for testing purpose)
- AI: continuous samples, samples to read: 30k, rate: 1k
2) I am using flat sequence structure as a way to making the start point of saving AI and AO data same, but wonders if this is a right method or if there is other better approaches. (I had an idea of using an internal clock with "1-sample mode" for sync, but this may not work at the high speed sine sweep like 5kHz, right?)
3) I just want to provide the sine sweep just once to the system and do not require the "reset" functionality implemented in the original sample program. I failed to remove this "reset" part as I did not fully understand the sample code. If I run the attahed, the generated AO signal is periodically provided to the system. Please give me any advice to modify the program as I want.
Thanks for your help and valualble time.
Attachments:
siso.vi ‏230 KB

Hi J. Kim,
To begin, you say you want to synchronize your analog input and output so that they start at the exact same time, yes? To achieve tight synchronization, you need to use DAQmx VIs instead of the DAQ Assistant. There's a good overview of DAQmx VIs here. There's also a document that deals specifically with synchronization in DAQmx here. Additionally, if you go to Help>>Find Examples in the LabVIEW example finder, you can see many other examples of acquiring data using DAQmx.
As for your configuration, you have your analog input DAQ assistant wired far before your analog output DAQ assistant in your sequence strucutre, so the dataflow of the program will cause the analog input DAQ Assistant to execute before the output. They cannot be in different sequences if they are executing simoultaneously, and I would not use flat sequences here except to start the two tasks in DAQmx. Where did you find this example?
Best,
Dan N
Applications Engineer
National Instruments

Multiple inputs and outputs using DAQmx VIs

Hello,
I am fairly new with the LabView programming language. I have a few training books that I have been reading, and I have been following online tutorials and reading the forums. However, I have come to a problem where I don’t see a clear solution. I am using LabView 2009 (9.0f3) and DAQmx VIs.
I am using a NI 9172 chassis PLC, with two 9201 AI cards, 9217 AI RTD card, 9472 DO card, 9263 AO card, and two 9237 AI Bridge cards.
I am reading eight analog inputs with the 9201 cards, two analog RTD inputs with the 9217 card, three digital outputs with the 9472 card, three analog outputs with the 9263, and eight analog inputs with the 9237 cards.
I wrote a simple program to test one digital output task, two of the analog output tasks, and a single analog input task. I put all of them in the same while loop, and it worked perfectly. However, when I add analog input tasks to the same loop, I get an error 200022. So I tested each sensor individually by changing the channel before each run. I searched error 200022 and found that this is because I cannot start another analog input task until the previous one ends. With this said, I don’t know how to acquire an analog voltage in the same task as an analog RTD voltage. Both inputs take different constants in the start task DAQmx icon.
Attached is my test program. It is titled “Test All”. This is the program I used to test the various sensors. I tested the input sensors one at a time, and it worked fine. A few tasks are written just to test functionality, and will be added to later. The data is only displayed on the screen. I will add triggers and data write to disk functions later. This program works now, but if I add more analog inputs, it will generate the 200022 error.
Can someone show me how to correctly write the code for multiple inputs and outputs using DAQmx? All training materials and tutorials I can find all show a single input or single output, not multiples of each. Thanks for looking.
-Randy
Attachments:
TestAll-NI.vi ‏32 KB

Hi RandyC,
The Knowledge Base article Using Different Types of DAQmx Global Channels in the Same Task goes into a little more depth of what Bryan is talking about, and it also includes some example code to help show what to do.
Hope that helps,

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!

SIngle reference for PXI 6509 Input and Output

Similar Messages

Maybe you are looking for