Analog delay

hi .. I have an analog signal (2.225Mhz) .. and I want to get it delayed by 0.2 ms
Is there any idea ???
Thnx in advance

Use the component search and search for the keyword "delay" in function field, there is a signal delay part in the database. Here is a knowledge base to show how to search for parts in the Multisim database:
http://digital.ni.com/public.nsf/allkb/7309A5CABC677296862577ED006EC99E
Tien P.
National Instruments

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I have CS5 production, and the Sb program, has an analog delay effect. I deleted the default setting, and it disappeared. Is there any way to re-install it?

I have CS5 production, and the Sb program, has an analog delay effect. I deleted the default setting, and it disappeared. Is there any way to re-install it?

Hi Thanks a ton.. so I downloaded Sb from Cs5 download options
and it won't take the serial number, from the box. I bought Cs5
in 2010 in the box from adobe. It lists Sb5 on the back
of the box, and it's in line, and works fine.. paid about 1000 dollars
for the software. But, the downloaded exe file will not accept the
serial number. The same serial number in my account for Cs5
production premium suite. I guess the new cloud community doesn't
respond well to BOX installers or previous customers. Thanks a lot for
your effort but, at 40$ a phone call, I'll suffer through without the default
analog delay option.. everything else runs smooth so.. have a nice
day up there in the cloud.
Best Regards,
R Daves

How can I delay analog signal?

I have an analog signal, the frequency is less than 1kHz. I want to get a delayed analog signal. Is it possible to realize analog delay generator using labview (daq card is PCI-6115)? I hope I could specify the delay time.
Any suggestion will be greatly appreciated!
Stephen

Hello Stephen,
It is possible to institute a delay an analog output. This is best done by specifying a trigger and then beginning the output when that trigger occurs.
Below is a link to an example program developed for LabVIEW that waits a specified number of seconds and then triggers analog output to begin. This code performs analog input as well, but for your program you could remove that portion. The trigger that is used is RTSI pin 0. After a specified period of time, the RTSI pin 0 is set to 1 and then the analog output begins.
Analog Input with Delayed Analog Output
Take a look at this example and
try to build from it for your code. If you have any further questions on this issue, let me know.
Regards,
Scott R.
Applications Engineer
National Instruments

What's New in Audition 3.0?

Don't have time now to get into how they help, but I would just like to say that there is MUCH improvment in Audition 3.0...I'd like to say this because I've heard lots of grumbling on other forums about 3.0 not being an upgrade at all over the previous versions 1.5 or 2.0, this is completely false, for me in particular the improvement in MT View (editing: collectively trimming, duplicate entire track and fading improvements) are indispensable.
and most people don't even mention any improvement aside from spectral view (which is pretty big in itself, isn't it?) spot healing etc...which I haven't even used yet.
I just want to say, they definetly made some good improvements, while they may not be overwhelming to some they are entirely useful in my situation and I think for most people they would come to see they are definite improvements if they used it long enough.
Getting started / New features
What’s new
Record and mix
Adobe Audition 3.0 is a powerful tracking and mixing application. Mix faster with new automatic crossfades, clip fade handles, and automation-editing improvements. Take full advantage of the latest hardware with multicore processor support and an optimized mixing engine.
VST plug-in manager
Quickly enable or disable specific VST plug-ins, optimizing performance. (See Enable VST effects.)
Auto crossfades and clip fade handles
Simply overlap clips to crossfade them, and adjust fade curves with on-clip handles. (See Fade or crossfade clips in a track.)
Improved multitrack editing
Efficiently edit sessions with these key enhancements:
Collectively trim and fade grouped clips. (See Grouping clips.)
Ripple-delete ranges of clips, instantly removing time gaps. (See Trimming and extending clips.)
Adjust selected ranges of automation points. (See Edit automation envelopes.)
Duplicate the contents of entire tracks, including clips, effects, and automation. (See Duplicate a track.)
Simultaneously view all input and output levels to comprehensively monitor a mix. (See Monitor levels.)
XML session support
Save sessions to XML format and other shared standards for multitrack applications. (See Save multitrack sessions.)
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Quickly output a session directly to Edit View, without first exporting a file. (See Create a single audio clip from multiple clips.)
Video previews for surround mixes
Watch a preview in the Video panel while adjusting mixes in the Surround Encoder. (See Previewing video.)
Create and arrange
Adobe Audition 3.0 offers powerful and extensive looping capabilities, as well as support for VST instruments, making it easy to create and arrange great-sounding music. Improved processing, including the high-quality Radius time-stretching engine from iZotope and numerous new effects, gives you infinite creative options.
MIDI tracks and piano-roll editor
Import, record, and edit MIDI, and output it through VST instruments or hardware synthesizers. (See Composing with MIDI.)
New effects
Explore creative sonic possibilities with Convolution Reverb, Analog Delay, Guitar Suite, and other new effects. (See Effects reference.)
Radius time-stretching from iZotope
Access industry-standard algorithms in the updated Stretch effect, as well as the File Info and Audio Clip Looping dialog boxes.
Bitmap audio images
Export spectral graphs for detailed editing in an image-editor like Adobe Photoshop®. Or, import visually-oriented graphics as source material for experimental sound designs. (See Spectral Bitmap Image (.bmp) and Import a bitmap image as audio.)
Improved CD ripping
Automatically import track information from your favorite CD database. (See Extract CD tracks with the Extract Audio From CD command.)
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Sort files by track number, or by the date they were opened or created. (See Change how files appear in the Files panel.)
Customizable workspaces
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Edit and master
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Organize and play marker ranges for live performance and broadcast. (See Creating play lists.)
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ah guh suk yuh dutty sket mudda bumbohole battybway!
It's a very helpful post because there is people who come to this forum when trying to decide to what recording software too purchase, and there's also people all over the internet who say that Audition is not better then Pro-Tools and more amateurish when in many regards it is more professional and has a better interface and GUI, especially if you're not doing too many tracks (less then 16 or perhaps more on a high end system). It's multi-track is easy and stays out of your way, has great automation, and some updated editing capabilities and has an excellent mbc that pro-tools does not have (maybe not excellent but very good)..and I believe pro-tools is a cpu hog doing certain things just like Audition is.
So people come to this site and I'm going to tell them the truth that it has been added to nicely since Adobe bought it and hopefully if people buy it will have another update to make the multi-track even better. It has a few bugs and downsides but so does pro-tools, I think the upsides are greater then pro-tools.
The only downsides are the midi is a bit lacking compared to other programs which have been in the midi-game longer, and that it does use alot of cpu for certain efx, and the multi-track is better in some ways but not as advanced in others in terms of sequencing and arranging, it could improve in that area, loops and stretching loops to match each other then being able to record how you bring the loops in and out in a sequence like ableton but in it's own way then it would rule the world forever.

Syncing video output to the specified lag time (ms) of your monitor

Hey guys and lassies, I've got a question that I've been thinking about a lot, and I've looked around the web without finding a good answer, so hoping to get edumacated in the subject here!
I do a lot of video work in Premiere Pro CS6 where I want to be absolutely sure that I have perfect (or as near perfect as possible) synchronisation between video and audio. I'm building a new editing rig now and I'm wondering about monitors. What has me pondering at the moment is the different lag times the various screens have. Some of the TN screens are really fast, and some of the various kinds of IPS-screens are a lot slower. Maybe I'll use a TN for working and an IPS on the side for previewing and getting colors right. Is there a way to make sure that whatever lag time your screen has, be it 4ms or 40ms that you know that what you're seeing and hearing is actually IN SYNC for real, and not just because it synchs with your monitor? Can this be specified in the output settings anywhere?
I know that we're only talking about a frame or so here or there, but it's still something I'm really curious about and think would be interesting to find out. I've got a bad case of perfectionists syndrome. Thoughts on the subject?

i dont know of any programs or devices that can tell the report the delay from an external monitor. i guess if you really wanted to find out you could video record some test footage onto a camcorder, then bring that into premiere and look at the sync sounds on the timeline vs the video frame to see how far out it is. you would need high fps to really get accurate though and still may not be a good way to figure this out...
most people cannot tell delay over 100ms. so if your monitors are 4ms or 40ms and that is the limit of total delay vs audio it should not be noticeable. if it is noticeable and the footage is in sync on the timeline (frame vs waveform), you can try to compensate with an audio vst delay plugin (described below). just load up some sort of syncpop test footage in premiere and eyeball it to adjust the delay, to get an acceptable sync in playback on your monitors.
you can add a delay vst to the master channel, set it to zero dry, 100% wet and set the delay by milliseconds to get the desired result. the analog delay that is built into premiere might work, just have to double click on its name after adding to get to the controls. i haven't tested this, but this should be easy enough for premiere to handle the delay vst. if you need to add delay to the video this route may not work. i think there are some audio delay vst that get into negative delay but requires the host application to accept it.

How can I test IRIG-B'a accuracy using Labview

We have a GPS source as reference, how can we test second GPS clock with IRIG-B output ? Can we by using Labview ?

Most timecode generators work by taking an external input (GPS, IRIG, PPS, or any of a number of other timecodes such as MILA, NASA36, IRIG A, etc) and outputting an on-time signal.
The most common systems now use a GPS receiver to steer an oscillator to some fixed frequency (10MHz is a favorite) and then use that steered oscillator to generate the outputs. The one PPS from a GPS receiver is not very smooth and jumps around easily a few hundred nano-seconds every PPS (at least they used to).
Time codes are usually done by generating an on-time digital signal in which the rising edge of the frame bits is co-incident with the PPS timing. The digital signal is then sent through some analog magic (op amps and whatever) which delays the output on time.
In digital signals, the rising edge is considered on time, in analog it is the zero-crossing.
Good systems will calibrate out the analog delay in the factory by moving back the on-time timing that is fed to the analog so that all on-time marks are co-incident.

Effects heard only on playback

I have Logic Express 9.1.8 with a Roland Cakewalk UA-25ex USB interface. When selecting effects from the library, i.e. electric guitar, crunch delay, crunch analog delay pedal I don't hear them as I play but only on playback after recording.
Is this normal and by design or should I be able to hear any of these effects? how would I set this up to hear all the time?

I found the answer to my problem. Under the core audio preferences software monitoring must be checked to hear the out put in real time. Hope this helps others as it drove me nuts for a while.

Seeking assistance with replicating sound effect.

Greetings one and all. I´m just a newbie looking for some help
As the title suggests. I would like assistance in replicating the sound effect(s) used in the following video in adobe audition : https://www.youtube.com/watch?v=7iexlGj4Lvg
I´ve tried effects such as chorus, flanger, phaser, reverb and that makes it sound better but it just lacks a certain ooomph that I just can´t put my finger on. Now after much head scratching and frustration I turn to you guys.
Thanks in advance.

Ah, so delay was the magic tool. certainly when I play around with it I can see the theory behind the sound effect. Whenever I use the delay I noticed I cannot adjust the feedback on it. I can however raise the feedback if I use the analog delay but then it sounds different. Any tips to make the the delay short and very strong like in the video? Perhaps using distortion to make the sound more powerful. Do you think the chorus effect is involved?

Compressor effect modes missing

In the training videos that come with CS3 Production Premium, when the compression effect is added to the rack, the pop up settings include a mode selection dialog. When I am editing audio from a type 2 AVI, I do not have that option. The tool only displays a slider for the amount of compression.
Where are the modes?

In the Effects panel, when you click the "fx" button and see a flyout listing the effects, the bottommost entry is another flyout labeled "Advanced." These consist of the same effects, but with full parameter options.
For example, the basic Analog Delay effect allows users to choose from the available presets and adjust the amount that the preset parameters are applied. The Advanced Analog Delay effect displays all of the available parameters to the user.
Durin

How to avoid delay during analog output generation by changing its frequency?

Windows XP
LabVIEW 7.1
PCI-6036E + BNC-2120
Hi,
I am going to create a vi to generate an engine speed sensor signal (a simple square wave with specific missed pulses, in my case 58 pulses “teeth” and 2 missed pulses “missed teeth”) as an analog output but in addition give me the opportunity to control parameters for example frequency online to simulate the engine speed changes during running that vi. For this purpose I have started with “Continuous Generation.vi” which is available in NI Example Finder under the following path:
Hardware Input and Output > Traditional DAQ > Analog Output > Continuous Generation.vi
Then I modified it towards above mentioned goal, all related vi s are attached. The main vi is: "Motor Signal Generator_1.12.vi"
At the first try it looks that it works properly but when have a look on that more accurately with Oscilloscope (fortunately I have a good one: Agilent 54621A – 60 MHz, 200 Ms/s) obviously there is a gap (delay or Jitter) whenever I change the engine speed. It is also attached in Signal generation_problem report.doc file.
Note: Small gaps are OK and related to predefined missed teeth but the big one is happened during changing engine speed.
As far as I understand it is related to the time which case structure in AO C-GEN sub-vi needs for AO reconfiguration each time after changing the engine speed (update rate). How can I get rid of this delay or gap during signal generation and generating completely continuous signal?
I have to mention that obviously I changed the frequency by changing the update rate. The other possibility is to change the number of updates in one period (refer to "generate arb frequency.vi" in NI site: http://sine.ni.com/apps/we/niepd_web_display.display_epd4?p_guid=B45EACE3E48F56A4E034080020E74861) which resulted in no delay however then I can not change the frequency continuously but step by step (for example jump from 5Khz to 2.5KHz immediately) and this can not pass to my application.
Any hint is appreciated.
regards
Attachments:
Signal_generation_NIsupport.zip ‏81 KB

Hi Roozbeh,
The following example will allow you to vary the pulse train frequency during run time.
Thanks,
Lesley Y.
Attachments:
GenDigPulseTrain-ChangingSpecs.vi ‏75 KB

How can I set up a delayed analog trigger on PCI 6115 DAQ

I have an S-Series PCI 6115 DAQ which I’m running with Labview. I’m using it to measure signals from an acoustic emission sensor and two force transducers. I’d like to set up a delayed analog trigger which will start acquisition on all three channels a period of time after a selected channel’s voltage exceeds a threshold.
Currently I’m using the AI Config VI in line with the AI Start VI and AI Read VI to capture data after a analog hardware trigger occurs. A software trigger probably wouldn't work because I have to sample my data at 10MS/sec. My setup works fine for triggering without any delay or skip counts. However, if I set the delay or skip count in the additional trigger parameter field of the AI start VI, there is no effect, and the device still starts capturing data immediately after the trigger is received. What is the cause of this, and how can I get around it?
Also, is it possible to sample the channels of a PCI-6115 DAQ at different rates? Right now, I’m sampling all my channels at 10MS/sec and throwing away data on all channels except one. However, this seems relatively slow and eventually I would like to attempt pseudo-real time control using my data.

rpursley8 is right about needing to get the counters involved if you want a hardware timed delay in your application.
Concerning whether or not you can sample at different rate, check this document out.
Sampling Different Channels at Different Rates with NI-DAQmx
Otis
Training and Certification
Product Support Engineer
National Instruments

How to combine Digital Output, a delay and Analog Input in a fast loop

I need to develop a process loop that runs at least at 250 Hz that performs a Digital output, than a delay of 50 microseconds and than an analog input of all the channels. All will be done using ATI MIO 64E3 card. Of course, the acquired data will be processed, displayed and saved. The loop will be running for several minutes until user stops it.

The fastest and most precise timing will occur if you use hardware timing. You can apply hardware timing to analog input on the E Series boards, but not the digital lines. Let's focus on the analog input first. Continuous waveform scanning uses a scan clock, which can be the board's internal one or an external one which you apply. If you want to scan all the channels 50 microseconds after a digital rising edge, then you need an external signal to signify that scan clock.
The E Series boards also have 2 counter/timers onboard that you can use for this purpose. You can set up a retriggerable pulse generation operation, where the counter receives a trigger and then on the user specifications, produces a pulse. You can have that route to the analog input scan clock.
The trigger signal for the counter is that digital pulse. As I mentioned earlier, there is no hardware timing for the digital lines on an E Series board. We do have other digital boards (653x family) that have hardware timed operations if precision is important. If you are satisfied with software's resolution (in the milliseconds), then you can call the E Series board digital function in a loop with a software timer. That digital line can route to the counter to act as the trigger.
So, on the programming side, you can have three separate and independent operations in parallel. One is for the digital function to output on that line every so often. Another is for the counter set at the retriggerable pulse generation. The last is for the analog input. I will describe this in terms of LabVIEW, but it can be done in a similar fashion with the NI-DAQ function calls or Measurement Studio.
The digital examples are in the LabVIEW >> Examples >> Daq >> Digital >> E-Series directory. The Generate Retriggerable Pulse example is in the LabVIEW >> Examples >> Daq >> Counters >> DAQ-STC directory. The E Series boards use the DAQ-STC timing chip.
Go to the LabVIEW >> Examples >> Daq >> anlogin >> strmdisk.llb directory and start with the Cont Acq to Spreadsheet File. This shows how to continuously acquire data and stream it to disk while displaying the data on a chart. Streaming to disk is the efficient way to save data while you are acquiring, as it eliminates the overhead of always opening and closing the file through the iterations of the loop. This saves to a file that can be opened by other applications (Excel, Word, etc.), but it is not as fast as writing to a binary file, which must be opened and read back through LabVIEW. However, for your ~250 Hz rate, it should be fine. Then, go to the LabVIEW >> Examples >> Daq >> anlogin >> anlogin.llb and look at the Acquire N Scans -ExtScanClk example. This shows how to apply the scan clock. Here, the AI Start that you saw in the previous example is replaced by 4 VIs (3 AI Clock Config's and the AI Control). Make those changes to the first example and then add a constant 0 to the AI Control parameter for total scans to acquire. That specifies the continuous operation. The File >> VI Properties >> Documentation menu item of the example describes the physical connections.
If you aren't using LabVIEW, use the NI-DAQ User Manual and the NI-DAQ Help file installed on your machine. You can look at your AT E Series User Manual at the http://www.ni.com/manuals pages for more information on the hardware. Also, if you want to route those signals internally on the board, you can find some entries in the KnowledgeBase at the http://www.ni.com/support pages.
Regards,
Geneva L.
Applications Engineering
National Instruments
http://www.ni.com/ask

Generate a delayed pulse with analog trigger

I want to generate a pulse with counter when the gate of counter receives a voltage value which is generated by analog output chanel,is this possible??
I am using PCI-6713
Thanks!!

You can generate a triggered pulse in a counter by sending a signal to the gate from the Analog output, as long as such analog output meets the operational contitions of the counter of the PCI-6713. Just make sure that the signal does not exceed 5V. The trigger on the gate pin will be taken as low from 0 to 0.8V and as high from 2 to 5V.
Thanks for contacting National Instruments!

Delayed repeated analog output

I have an AT-MIO 2 DAQ card. I am using the FIFO-buffer for outputting
to the two outputs. The content of the Buffer stays the same and has to
be outputted multiple times. After every output the output should wait a
time and then restart?
How do I do that ...
WFM_ClockRate(dev_nr,group,1,-1,1000,1) does not seem to work .....

You can use software timing or hardware timing. Since you'll be running on a Real-Time operating system, software timing will be much better than if it were running on Windows. You also have the option of configuring an interrupt to "fire" and execute code whenever you want.
I hope this helps,
Kevin S.
Applications Engineer
National Instruments

Creating a Time Delay Between 2 Waveforms

Hi,
Please bear with me while I explain what I’m trying to do :-) Basically, in the attached LabView file (“flow vis + trigger-m.vi”), I am generating 2 signals: a sinusoidal waveform (used to drive an actuator) and a square waveform (used as a TTL signal to trigger image capture on a CCD camera). Currently when I run the program using LabView 7, both signals are started simultaneously. I would like to introduce a time delay that can be manually specified in the VI, such that when the program is started (i.e. at time t0), the sinusoidal waveform also starts (i.e. at time t0) before the square waveform (i.e. at time t0 + delay time). In a practical sense, this means allowing the actuator to run for several seconds before capturing any images.
It sounds like it should be quite straightforward to do, but I’m not so sure how to progress. I have begun to modify the original VI (“flow vis + trigger-m_2.vi”, as attached), but would be much appreciative of any help or suggestions on how I can complete the VI to meet the above requirement.
Many thanks,
Mark
Attachments:
flow vis + trigger-m.vi ‏939 KB
flow vis + trigger-m_2.vi ‏895 KB

Hi Mark,
I forgot that all analog waveforms generated needed to have the same number of samples in a single task. By introducing a delay into your square wave, I effectively added samples that waveform. So the same number of samples added to the square wave has to be added to your sine wave. I added code that extends the number of samples of your sine wave (# delay + square wave samples). Let me know if this works for you.
Regards
Way S.
NI UK Applications Engineer
Attachments:
flow vis + trigger-m70.vi ‏872 KB

Analog delay

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