Measuring lag between input and output sine waves...
Hi,
I am using a Labview VI to control an array of servo motors, each receiving a sine wave input (with different phase offsets for each servo). I'm using a NI USB-6229 DAQ system to monitor their movement via a wire that I've soldered onto each servo's potentiometer. I've written another VI that recieves both the initial sent signal and the returned DAQ signal and plots both on a waveform chart. Both signals show up on the plot as expected, with an obvious phase shift between them. I'm having problems accurately measuring the phase shift, and I'd rather have an automatic method that gives accurate numerical values instead of a more qualitative visual method.
So far I've tried to use triggers and peak to peak comparisons, but I've been having problems getting results in real time. My incoming signals are time stamped, if that helps. Is there a quick and easy method to extract phase shift data from two sine waves on a waveform chart?
Thanks for the help...
Since I work with lock-in amplifiers I came up with this. I guess it will work in your case. This method has some limitations. It will work up to 180 degree. It also measure absolute phase lag. You can not say if the signal is leading or lagging. But in your case it will always be lagging. I posted a VI showing the method in another discussion. Here is the link. http://forums.ni.com/ni/attachments/ni/170/409516/1/demo_v2.vi. Since your DAQ unit do not have simultaneous sampling you will get a small phase error due to inter-channel delay. This is also true if you use the FFT method. What is your signal frequency and which samplerate do you use?
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)
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CFP-AIO-610 electrical isolation between input and output channels
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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. -
Difference between input and output values
I am using NI DAQ 6024E and SCB 68.
I have shorted one anlog output and input channel
in the SCB 68 board itself.(differential connection). that is whatever ouput voltage
i am giving at o/p channel 22 and 55 should come at i/p
channel 68 and 34. But when i read the values in my VI,
the input voltage is higher than o/p. why is the difference.
I used default buffer size 1000 and AI Single scan and AO Single Update.There doesn't seem to be a problem with your pin connections.
The SCB-68 has a temperature sensor that can be used as cold-junction compensation for thermocouple measurements. If the temperature sensor is enabled, it will use channel 0 of the DAQ device.
Since you are connecting to channel 0 of the 6024E, I would make sure you have the temperature sensor of the SCB-68 disabled. You can disable the temperature sensor by using the switched of the SCB-68. The User Manual of the SCB-68 is linked below and provides the information on how to disable the temperature sensor (page 2-4).
SCB-68 68-Pin Shielded Connector Block User Manual
http://digital.ni.com/manuals.nsf/webAdvsearch/74C86ADEF0E4813F86256C84007CB3AB?OpenDocument&vid=niwc&node=132100_US
I hope t
his helps.
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Todd D. -
How to switch USB-6008 between digital input and output modes
Hi I have been following the examples of setting a specific port to either inout or output using the DAQmxCreateDIChan() or DAQmxCreateDOChan() calls. What I now want to do, is switch betwen inout to output mode and back again.
DAQmxCreateDOChan()
DAQmxWriteDigitalLines()
// do something
//switch to input
DAQmxReadDigitalU32()
// switch back to output
I can't seem to find any calls or discussions on this.
rjmillerHi RJM,
If you want to be able to switch your digital line between input and output, then you will need to use the Tristate Property. There is another discussion forum with more detailed information about using this property and I believe that you have already seen it. I am going to post the link in case anyone else wants to see the other forum as well: Tristate Property Discussion Forum. Reddog's post is very informative.
Regards,
Hal L. -
How to get the sine wave sample of the input and output in the same time?
I discovered that the samples were acquired did not in the same time in my vi.Could you tell me how to get the sine wave sample of the input and output in the same time?
Output:9263;
Input:9201
Attachments:
singal requency.vi 104 KBHello Yun_Zhu,
Your VI looks like you started from an example, and you are on the right track.
To clarify are you trying to synchronize the input and the output waveforms? If you you will need to share a time base and a start trigger. Check out the synchronization examples from LabVIEW
Anthony F.
Product Marketing Engineer
National Instruments
Attachments:
HIODAQmxSyncMultiDev.png 70 KB -
Synchronize input and output tasks to start at the same sample point [C++ NI_DAQmx Base]
I'm trying to initiate the analog input and output streams to start reliably at the same sample. I've tried triggering the output from the start of the input using the following code [NI-DAQmx Base 2.1 under Mac OS X with an M-Series multifunction board]. It compiles and runs, but gives an error message at the call to "DAQmxBaseCfgDigEdgeStartTrig". Any suggestions about synchronized I/O on this platform?
#include "NIDAQmxBase.h"
#include
#include
#include
#define DAQmxErrorCheck( functionCall ) { if ( DAQmxFailed( error=( functionCall ) ) ) { goto Error; } }
int main( int argc, char *argv[] )
// Task parameters
int32 error = 0;
TaskHandle inputTaskHandle = 0;
TaskHandle outputTaskHandle = 0;
char errorString[ 2048 ] = {'\0'};
int32 i;
time_t startTime;
// input channel parameters
char inputChannelList[] = "Dev1/ai0, Dev1/ai1";
float64 inputVoltageRangeMinimum = -10.0;
float64 inputVoltageRangeMaximum = 10.0;
// output channel parameters
char outputChannelList[] = "Dev1/ao0, Dev1/ao1";
char outputTrigger[] = "Dev1/ai/StartTrigger";
float64 outputVoltageRangeMinimum = -10.0;
float64 outputVoltageRangeMaximum = 10.0;
// Timing parameters
char clockSource[] = "OnboardClock";
uInt64 samplesPerChannel = 100000;
float64 sampleRate = 10000.0;
// Input data parameters
static const uInt32 inputBufferSize = 100;
int16 inputData[ inputBufferSize * 2 ];
int32 pointsToRead = inputBufferSize;
int32 pointsRead;
float64 timeout = 10.0;
int32 totalRead = 0;
// Output data parameters
static const uInt32 outputBufferSize = 1000;
float64 outputData[ outputBufferSize * 2 ];
int32 pointsToWrite = outputBufferSize;
int32 pointsWritten;
for( int i = 0; i < outputBufferSize; i++ )
outputData[ 2 * i ] = 9.95 * sin( 2.0 * 3.14159 * i / outputBufferSize );
outputData[ 2 * i + 1 ] = -9.95 * sin( 2.0 * 3.14159 * i / outputBufferSize );
// ------------------- configure input task -----------------------
DAQmxErrorCheck ( DAQmxBaseCreateTask( "", &inputTaskHandle ) );
printf( "Created input task\n" );
DAQmxErrorCheck ( DAQmxBaseCreateAIVoltageChan( inputTaskHandle, inputChannelList, "", DAQmx_Val_RSE, inputVoltageRangeMinimum, inputVoltageRangeMaximum, DAQmx_Val_Volts, NULL ) );
printf( "Created AI Voltage Chan\n" );
DAQmxErrorCheck ( DAQmxBaseCfgSampClkTiming( inputTaskHandle, clockSource, sampleRate, DAQmx_Val_Rising, DAQmx_Val_ContSamps, samplesPerChannel ) );
printf( "Set sample rate\n" );
// ------------------- configure output task -----------------------
DAQmxErrorCheck ( DAQmxBaseCreateTask( "", &outputTaskHandle ) );
printf( "Created output task\n" );
DAQmxErrorCheck ( DAQmxBaseCreateAOVoltageChan( outputTaskHandle, outputChannelList, "", outputVoltageRangeMinimum, outputVoltageRangeMaximum, DAQmx_Val_Volts, NULL ) );
printf( "Created AO Voltage Chan OK\n" );
DAQmxErrorCheck ( DAQmxBaseCfgSampClkTiming( outputTaskHandle, clockSource, sampleRate, DAQmx_Val_Rising, DAQmx_Val_ContSamps, samplesPerChannel ) );
printf( "Set sample rate\n" );
// trigger output when input starts
DAQmxErrorCheck ( DAQmxBaseCfgDigEdgeStartTrig( outputTaskHandle, outputTrigger, DAQmx_Val_Rising ) );
printf( "Set output trigger\n" );
// ------------------- configuration -----------------------
// write output signal
DAQmxErrorCheck ( DAQmxBaseWriteAnalogF64( outputTaskHandle, pointsToWrite, 0, timeout, DAQmx_Val_GroupByScanNumber, outputData, &pointsWritten, NULL ) );
printf( "Write output signal\n" );
// set up input buffer
DAQmxErrorCheck ( DAQmxBaseCfgInputBuffer( inputTaskHandle, 200000 ) ); // use a 100,000 sample DMA buffer
// initiate acquisition - must start output task first
DAQmxErrorCheck ( DAQmxBaseStartTask( outputTaskHandle ) );
DAQmxErrorCheck ( DAQmxBaseStartTask( inputTaskHandle ) );
// The loop will quit after 10 seconds
Dr John Clements
Lead Programmer
AxoGraph ScientificHi Michael,
First of all, thanks very much for taking the time to investigate this problem! Much appreciated.
You asked for "an actual error code you got and any description that is given". The full output from the program that I posted earlier in this thread is appended to the end of this message. In summary, following the call to...
DAQmxErrorCheck ( DAQmxBaseCfgDigEdgeStartTrig( outputTaskHandle, outputTrigger, DAQmx_Val_Rising ) );
... with ...
char outputTrigger[] = "Dev1/ai/StartTrigger";
...the error message is ...
DAQmxBase Error: Specified route cannot be satisfied, because the hardware does not support it.
You asked "specifically which M series device you are using"? It is the PCIe 6251 (with BNC 2111 connector block). I'm testing and developing on an Intel Mac Pro (dual boot OS X and Windows XP).
You asked for "the location you pulled the code from". Here it is...
http://zone.ni.com/devzone/cda/epd/p/id/879
...specifically from the file "Multi-Function-Synch AI-AO_Fn.c".
I adapted the NI-DAQmx calls to their NI-DAQmx Base equivalents.
Finally, you asked "Is the trigger necessary, or do you just need to know that the measurements are running on the same clock?". I believe that some kind of sychronized trigger is necessary in my situation (correct me if I'm wrong). Timing is crucial. Say I initiate an analog output stream that delivers a voltage command step 5 ms from the onset. I need to record the response (analog input stream) so that its onset is accurately aligned (synchronized) at 5 ms. A typical recording situation would stimulate and record a short data 'sweep', then wait for the (biological) system to recover, then stimulate and record another short sweep, and repeat. I need all the recorded sweeps to align accurately so that they can be averaged and analyzed conveniently.
I definitely do not want my customers to rely on an expensive external TTL pulse generator to initiate and synchronize each 'sweep'. That would effectively eliminate the cost advantage of an NI board, as well as adding unnecessary complexity in setup and use. It would be a show-stopper for me.
It seems perverse, but would it be possible to use a digital output channel connected directly to a digital input chanel to trigger the input and output streams?
Regards,
John.
Full output from test program. Compiled with gcc 4 under OS X...
[Session started at 2007-05-23 14:17:01 +1000.]
LoadRuntime: MainBundle
CFBundle 0x303cc0 (executable, loaded)
_CompatibleWithLabVIEWVersion: linkedAgainst: 08208002
_CompatibleWithLabVIEWVersion: result= false, mgErr= 1, theActualVersion= 00000000
_CompatibleWithLabVIEWVersion: linkedAgainst: deadbeef
_CompatibleWithLabVIEWVersion: Reseting Linked Against
_CompatibleWithLabVIEWVersion: linkedAgainst: 08208002
_CompatibleWithLabVIEWVersion: result= true, mgErr= 0, theActualVersion= 00000000
_CompatibleWithLabVIEWVersion: linkedAgainst: 08208002
_CompatibleWithLabVIEWVersion: result= true, mgErr= 0, theActualVersion= 00000000
com.ni.LabVIEW.dll.nidaqmxbaselv
CFBundle 0x313760 (framework, loaded)
{type = 15, string = file://localhost/Library/Frameworks/nidaqmxbaselv.framework/, base = (null)}
Amethyst:Library:Frameworks:nidaqmxbaselv.framework
2007-05-23 14:17:02.248 test-ni[4445] CFLog (21): Error loading /Library/Frameworks/LabVIEW 8.2 Runtime.framework/resource/nitaglv.framework/nitaglv: error code 4, error number 0 (no suitable image found. Did find:
/Library/Frameworks/LabVIEW 8.2 Runtime.framework/resource/nitaglv.framework/nitaglv: mach-o, but wrong architecture)
CFBundle 0x1751fdc0 (framework, not loaded)
Created input task
Created AI Voltage Chan
Set sample rate
Created output task
Created AO Voltage Chan OK
Set sample rate
DAQmxBase Error: Specified route cannot be satisfied, because the hardware does not support it.
test-ni has exited with status 0.
Dr John Clements
Lead Programmer
AxoGraph Scientific -
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! -
Synchronisation of input and output using myDAQ
Hello!
I am currently using the NI myDAQ and use both analoge output. and both analoge inputs. The sampling rate needs to be as high as 200k Hz. (Also within the while loop I do quite a lot of data manipulation and saving into text files.) My vi works generally fine, but I still have to problems:
1. How can I ensure that the read and write tasks are synchronised? I found out that it is not possible to use other trigger than the sample clock for write and read tasks using the myDAQ. (Is this right?) So as a workaround for me it is not important that the tasks start at the same time, but I want to ensure that the delay between "Start Write Task" and "Start Read Task" always is the same. Is this possible? And how can I realise it?
2. If I want to generate a non periodical signal that is not constant, how do I best generate this? At the moment I use a Sine Waveform.vi with a frequency of 0 and I change the offset input with every loop. I also tried to use the FormulaWaveform.vi but with this my produced waveform seemed to stop after completing the "number of samples in the waveform". But the sampling info must be the same for both AO channels I use, so I cannot simply increase this number.
Thanks a lot for any help! Jules
Attachments:
MyDAQReadWrite.vi 72 KBhi Jules,
If you only want to ensure that the delay between "Start Write Task" and "Start Read Task" always is the same, you can realize it with a simple while loop (with Wait Until Next ms Multiple).
But if you want to synchronize for example both analog input, you can see how it works in different examples. Following Path is shows you how to find these examples:
In LabVIEW -> Help -> Find Examples... Brows(Brows according to Task): Hardware Input and Output -> DAQmx -> Synchronization -> "Analog Input Synchronizantion.vi"
To generate a non periodical signal that is not constant, you can change the offset input every loop. If you want to manipulate the offset while running you can also use local variables in the loop.
In this case you could manipulate for example with a slide control.
Regards,
grroesis -
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!
-allenYes 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. -
FINALLY INPUTTING and OUTPUTTING Annotations!
Ok, before everyone thinks that this is a very bad solution, I have to tell that I`m no programmer and my knowledge of PostgreSQL, Ruby or any other language is very poor.
With useful help from Jamie Hodge, fbm and Nicholas Stokes (mainly)
I could manage to write a command for inputting and outputting from Final Cut Server Annotations.
So lets go to the fun part:
INPUTTING:
1- Create a Response called "Annotations IN" (or whatever you want):
a - Reponse Action: "Run an external script or command"
b - Run Script > *Commnad Path: /Library/Application Support/Final Cut Server/Final Cut Server.bundle/Contents/Resources/sbin/fcsvr_run
Command Parameters: psql px pxdb -c "COPY pxtcmdvalue FROM '/FCSRV/annotation-in.txt' USING DELIMITERS '|';"
2 - Create a poll Watcher with name: "Watch for Annotations IN"
a - Enable: true
b - Monitor Address: Chooses a Device (create a new one or use a tmp one) and path to where you`ll going to put a txt file with the annotations.
c - Response List: Choose the Response you created "Annotations IN" in my case.
d - Event Type Filter: Created, Modified
e - Poll Watcher > Listing Frequency: 2 (or any number of seconds you feel like it).
Listing multiple: 2
Wildcard include Filter: *.txt (or any custom extensions you want)
3 - Create a txt file and use this as a template:
{ASSET_ENTITYID}|1527|{TCIN}/(30000,1001)|{TCOUT}/(30000,1001)|{Annotation}|{USE RID}|{DATE}
Where:
{ASSET_ENTITYID} = Is the entityid of your asset. You can find what number it is by issuing:
/Library/Application\ Support/Final\ Cut\ Server/Final\ Cut\ Server.bundle/Contents/Resources/sbin/fcsvr_run psql px pxdb -c "SELECT pxmdvalue.entityid, pxmdvalue.value AS asset_name FROM pxmdvalue INNER JOIN pxentity ON pxentity.entityid = pxmdvalue.entityid WHERE pxmdvalue.fieldid='1543' AND pxentity.address LIKE '/asset/%';"
This will output ALL your assets, so if you know the name or want to parse the name you can use:
/Library/Application\ Support/Final\ Cut\ Server/Final\ Cut\ Server.bundle/Contents/Resources/sbin/fcsvr_run psql px pxdb -c "SELECT pxmdvalue.entityid, pxmdvalue.value AS asset_name FROM pxmdvalue INNER JOIN pxentity ON pxentity.entityid = pxmdvalue.entityid WHERE pxmdvalue.fieldid='1543' AND pxmdvalue.value='ASSETNAME' AND pxentity.address LIKE '/asset/%';"
Where in ASSETNAME you`ll have to put your Asset Name without extension.
{TCIN} and {TCOUT} is, of course, the TC`s points. In the form of: HH:MM:SS;FF
{Annotation} is the commentary.
{USERID} (in my case was 1)
{DATE}: This one is tricky. My example is 2009-03-15 19:31:15.839795-03
So is in the form YYYY-MM-DD HH:MM:SS.????? I really don`t know the rest. Could be milliseconds?
Of course one can write a script to translate everything from a txt file like:
ASSETNAME | TCIN | TCOUT | ANNOTATIONS | USER
But as I`ve said I`m no programmer
Ok.. now the time for the OUTPUT:
The command-line is:
/Library/Application\ Support/Final\ Cut\ Server/Final\ Cut\ Server.bundle/Contents/Resources/sbin/fcsvr_run psql px pxdb -c "SELECT pxmdvalue.value AS Asset_NAME, pxtcmdvalue.value, pxtcmdvalue.begintc, pxtcmdvalue.endtc FROM pxmdvalue INNER JOIN pxtcmdvalue ON pxmdvalue.entityid = pxtcmdvalue.entityid WHERE pxmdvalue.value='ASSETNAME';"
Where ASSETNAME is the Asset name without the extension.
Or issuing the following to OUTPUT ANNOTATIONS from ALL assets:
/Library/Application\ Support/Final\ Cut\ Server/Final\ Cut\ Server.bundle/Contents/Resources/sbin/fcsvr_run psql px pxdb -c "select * from pxtcmdvalue;"
Adding "> /PATHTO_WHERE_IN_WANT/ANNOTATIONSOUTPUT.TXT" at the end will put all output into a txt file.
It`s possible (in theory) to:
1- Create a boolean md field in FCSRV called "EXPORT Annotations" (don`t choose lookup)
2- add or create a md group called "Export Annotations" and add the above md field to it (don`t choose lookup)
3- Add "Export Annotations" md field to Asset Filter md group
4- Make a Response for Running external command. Command path: /Library/Application Support/Final Cut Server/Final Cut Server.bundle/Contents/Resources/sbin/fcsvr_run
Command Parameters: psql px pxdb -c 'SELECT pxmdvalue.value AS Asset_NAME, pxtcmdvalue.value, pxtcmdvalue.begintc, pxtcmdvalue.endtc FROM pxmdvalue INNER JOIN pxtcmdvalue ON pxmdvalue.entityid = pxtcmdvalue.entityid WHERE pxmdvalue.value=[FileName];' > ~/Desktop/ann-out.txt
(I`m having problem with this, it doesn`t work).
5- Make a Subscription that if Export Annotations modified = true, trigger if changed and trigger the Response above.
6- Add exporting annotations md group to Media md set.
In theory it`s possible to modify the FinalCutServerIntegrationSample get and input annotations instead of adding another "comment" field to md group list.
Few!
Ok so please help beautify this out!
This will be very useful for a lot of people.... We know that it`s only a matter of time to FCSVR have this function built-in... but this "time" could be years.
So let`s start ourselves!
Thank you very much!!
Regards!AlphaBlue wrote:
jverd wrote:
What were you hoping for? Someone reading your mind for what exactly you need, and handing you a giftwrapped answer, without you doing any work or clearly communicating a specific problem? This site doesn't really work that way.Fair enough. I'm asking for insight into how to input from and output to a StarOffice Spreadsheet into/from a Java program. (Think java.io except with spreadsheet files instead of .txt files.) I already answered that question.
I need to accomplish this without the use of a community-created library.That's a bizarre requriement. Why?
>
Okay, [here you go|http://lmgtfy.com/?q=communication+between+StarOffice+Spreadsheets+and+Java].
If you don't have any knowledge or experience, on the matter, please refrain from directing me to a google search. I assure you, I have already performed such a task.How would I know that.
But okay, let's say I know that. Let's say you bothered to point out that you've already done that. Further, let's say I do have knowledge of the subject at hand. Maybe I'm an expert. Maybe I wrote Star Office and an open source library (which wheel for some reason you must reinvent). You are assuming that I also have psychic powers, so that I can read your mind and know exactly what you've read so far and exactly what parts of your very broad and vague question said reading did not answer.
In short, you have received answers commensurate with your questions. -
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 KBHi 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 -
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 KBHello,
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 -
R12 - Input and Output Tax Classification Codes in Invoices Workbench
Hello,
After upgrade from 11i to R12, in invoices workbench, the Tax Classification Code LoV (invoice line field) contains taxes that formerly were both input and output.
In the E-Business Tax Implementation guide says that :
Release 11i tax codes and tax groups migrate to E-Business Tax as tax classification
codes. Payables and Purchasing tax codes migrate as tax classification codes under
ZX_INPUT_CLASSIFICATIONS. Receivables and Projects tax codes migrate as tax
classification codes under ZX_OUTPUT_CLASSIFICATIONS.
In invoices workbench form (APXINWKB), I looked in the record group for the LoV and the values are all from the same view zx_input_classifications_v.
SELECT lookup_code, meaning, description
FROM zx_input_classifications_v
WHERE lookup_type = 'ZX_INPUT_CLASSIFICATIONS'
AND org_id IN (:inv_sum_folder.org_id, -99)
AND enabled_flag = 'Y'
AND nvl(:inv_sum_folder.invoice_date, sysdate)
between start_date_active and nvl(end_date_active, :inv_sum_folder.invoice_date)
UNION
SELECT lookup_code, meaning, description
FROM zx_input_classifications_v
WHERE lookup_type = 'ZX_WEB_EXP_TAX_CLASSIFICATIONS'
AND org_id IN (:inv_sum_folder.org_id, -99)
AND enabled_flag = 'Y'
AND nvl(:inv_sum_folder.invoice_date, sysdate)
between start_date_active and nvl(end_date_active, :inv_sum_folder.invoice_date)
It follows that some formerly Payables and Purchasing tax codes were migrated as 'ZX_INPUT_CLASSIFICATIONS' or 'ZX_WEB_EXP_TAX_CLASSIFICATIONS'.
Anybody knows how is the 11i tax code data migrated to R12 ? Is this the intended behavior ?
ThanksI'm not a functional expert but I would suggest reviewing the following docs and see if it helps.
How To Update EBusiness Tax Lookup Codes? (Doc ID 466578.1)
Payables - Invoices - Invoice Lines Tax Classification LOV duplicating entries (Doc ID 1265315.1)
Upgraded Tax Classification Codes Not Enabled For Iexpenses (Doc ID 851607.1)
Thanks,
Hussein -
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 KBOne 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 -
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?
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