Synchronize AO channels in dotnet

Hi,
This is probably an easy one. I am simpy trying to synchronize 4 analogue output channels from my DAQmx card and if I write them as seperate tasks (i.e. a task for each channel) then write the output ramp waveform it will simply write the ramp from one channel after the other channel in the order for which the lines appear in the code - the lines of code look as follows: aowriter.WriteSingleSample(true, ramparray);
I realize that I could combine multiple channels in one task and then call the WriteMultiSample() method and then trigger this externally via a digital trigger but this is turning out to be problematic.
I guess I am wondering if there is a simple solution where there is a setting which can be changed such that each analog write task can be executed without waiting for the previous ramp applied to a different channel to finish
thanks
Jonathan

Hi Jonathan,
Sounds like you are trying to simply to a triggered AO operation on one board
that outputs several different signals on your different AO channels.
Doing simultaneous updates for multiple AO channels is most definitely not a
problem. All you have to do is know which data you are sending into which
channels. In the case that you are using you will need to write an array
of data to your AO task.
One thing to keep in mind is that you need to make sure that you only use one
task of one type for one board. In this case, since you are using
one board, then you should just have one AO task and then write the data
appropriately.
It sounds like all you really need to do is merge your arrays of data so that
the proper points are output at the right time. Once you get the data
going into the board right it won't have a problem outputting everything in the
manner that you desire.
Regards,
Otis
Training and Certification
Product Support Engineer
National Instruments

Similar Messages

Alazar count edges synchronization

Hello,
We have an Alazar AT9350 of which the acquisition has to be synchronised with the acquisition of a Count-Edges channel on an NI-6259 DAQ board. Running them synchronously is easy, for we run the Alazar board with a trigger signal (50KHz) that we can route to the NI board as well. However, starting them synchronously poses more of a challenge. The original solution was to input the trigger on the NI-board and reroute it when the count edges VI starts. However, the board needs one counter to create the sample clock (or receive the trigger signal), and one counter to output the signal. This leaves no counter for the Count-Edges channel (our NI board has only two counters).
I have come up with a possible solution for synchronisation, but it is a bit ugly. I output the task just after the start task VI to an indicator, and create a local variable. I connect a read version of the local variable to the acquisition while loop of my alazar board. My assumption: because the while loop has to wait till all data flows have presented data to him, the while loop must start synchronously with the start task. The actual program is much bigger, that's why I'm using a local variable here. A direct wire is also possible, but it would look even uglier.
I have put a snapshot of a simplified version of my program as an attachment. I realise I am not actually saving any data or outputting what the Count-Edges reads, of course this is different in my real version.
Is my method viable or should I use a different method? (or perhaps it is viable, but I should still use a different method).
Solved!
Go to Solution.
Attachments:
labview program.JPG ‏114 KB

Dear GerdW,
Thanks for your suggestions.
[quote]Probably not "real time"… When you use DAQmx to read your input data you can synchronize input channels. Have you looked in the example VIs for that case? [\quote] I use an NI board with DAQmx drivers to acquire the Count Edges. I use an Alazar board with their own SubVI's for the other acquisition. Therefore I cannot use the DAQmx synchronisation (which would've been quite easy indeed). And indeed, by "real-time", I of course mean that the amount of data I process per second is the same as the amount of data that I acquire per second .
I changed my example, hope I finally understood you well. I added a wait function to the loop that sends the notification. Otherwise the notification is sent on the same time, but there is still no synchronisation because the notifier is retrieved upon reaching the loop (and it essentially does nothing). To make sure I know when I start the two loops synchronously, I will add a high-resolution relative seconds to both loops and subtract them in another loop. If there is a better way to test synchronisation I'd happily take any suggestions. (Though perfect synchronisation is the best, one or two ticks off won't do much harm).
Attachments:
MinimalExampleWhileLoopSynchronisation.PNG ‏14 KB

Need advice on best way to acquire multiple vibration waveforms from a PXI-4472 board

I'm using a PXI-4472 board to acquire vibration data from 8 proximity probes, each with different scaling. Confused as to best approach to achieve this given all the options with MAX, LabView and its Sound & Vibration toolset scaling. Reading other posts, it appears that 24 bit acquisition also requires some special treatment.
This acquisition needs to be done continuously.
As an additional side note, I will eventually need to synchronize these channels with the channels on 2 other 4472 boards.
Any advice on how to use MAX in combination with LabView's AI CONFIG, START, READ, CLEAR and scaling
for this application would be greatly appreciated.
All the best,
Hunter

Hello Hunter --
In regards to your questions...
For scaling, I would recommend using either MAX or the Sound and Vibration Toolset (SVT) scaling, but not both together. The SVT is a bit more flexible, and in general I prefer that. One other thing to keep in mind about MAX is that virtual channels for accelerometers and microphones provide an option to control IEPE excitation, but in fact this can only be set from LabVIEW for the 4472.
24 bit acquisition does not require any special treatment if you are only reading the scaled (i.e. voltage or engineering unit) data. If you use the binary array option for AI read, you will only receive the top 16 bits.
If you use the SVT scaling, you should not worry about MAX at all. If you you do define virt
ual channels in MAX for scaling, you should simply wire a channel array with those channel names into AI Config. Other than this, what you do in MAX should not change how you write the LabVIEW code.
Please see the attached example for synchronizing PXI-4472s.
Hope this helps!
Bryan
Attachments:
PXI_4472_Continuous_Synchronized_AI.vi ‏156 KB

Phase-measurement, PXI-4461

Hi
I would like to do a hydroacoustic measurement with a common signalsource and two hydrophones. The hydrophones are mounted in parallell (same direction) at a fixed distance, - and is directed towards the signalsource with a given angle. This means that if I give an impulse from the signal source, the soundwave will reach the two hydrophones in the same moment IF the angle is zero. If the angle is larger than zero, the soundwave will reach the nearest hydrophone first. I would then like to find the delay between "signal-aproach" on the two hydrophones, because this will give me the actual angel to the signalsource in the actual plane. Frequency is 20-100kHz, ane I need a lot of "dynamic space". Is 4461 the right alternative, and how acurate can I do this angle-measurement ? Can I do it this way at all ??
Svein Erik

Hi Svein,
Since you are wanting to find out time delay between two signals you will definitely want synchronization. So any of the DSA devices should work well for that since they use one ADC per channel and all the channels are synchronized. Additionally there are some S-series boards which can sample a little faster up into the MHz range. These S-series boards are lower resolution: 14-18 bits instead of 24 bits, however they are simultaneously sampled so you can maintain tight synchronization between channels. http://sine.ni.com/nips/cds/view/p/lang/en/nid/10955
I'm not sure how to quantify "a lot of dynamic space," but I will say that a 446x family is going to be one of the best options when it comes to dynamic range both because it has a high dynamic range and it has many input ranges. I don't know how much you really need however, so that's difficult to say. There are other higher channel count offerings in the DSA family that may have the functionality you want. http://sine.ni.com/nips/cds/view/p/lang/en/nid/12051
If you are considering eventually doing three-dimensional algorithms, then I suppose you will need more than two input channels. The 4462 shares the same architecture as a 4461 but has four input channels instead of two inputs and two outputs.
How accurately can you make the measurement? The DSA family generally publishes an "AI Interchannel Phase Mismatch" spec. It is on page 10 of the 446x specifications http://www.ni.com/pdf/manuals/373770j.pdf . So depending on which input range you want to use, you could have as good 0.08degrees or better. But it would depend on your application.
Hope that helps.
Daniel
Conditioned Measurements Hardware

Write Digital Out on 6133 board using RT

I would like to right a simple VI that writes data to digital outputs of a 6133 board on a SBC using Labview RT.

Steven,
I have made good progress on reading data from the 6713 board. I am now turning my attention to writing data to the analog out on the 6713 board. I have about 80000 data points that I would like to send to the analog out channel on the 6713 board. The data points have been prepared as unsigned integers in the range of 0 to 4095, where 2048 is to represent 0 volts. I would like to use DMA to send the data from the SBC to the 6713 board. The DAQmx Create Virtual Channel forces a selection of either voltage or current. If voltage is selected for the analog out, the range defaults to -10 and +10 volts. What VI or VIs do you recommend sending this data? The sample rate is about 600 kHz, which means the 80000 data points represent a waveform of about 133 ms in length. I also need to synchronize multiple channels with the identical waveform.
Thanks for your help,
solitude

Generate and synchronize 6 digital channels with differents phases/frequencies/duty cycles/...

Hello,
For some kind of tests we need to generate, with a 6534 card, 6 signals.
The 6534 has 32 digital E/S and can go up to 20MB/s !
Each channel must be independant so that it can have it's own frequency and duty cycle. But for exemple the second channel can be delayed in reference at first channel. I put an image so that you can understand better what i try to do (3 signals are shown but six are needed).
I see a lot of things on forum that are close to my probleme, but none seems to be as advanced as mine.
http://sine.ni.com/devzone/cda/epd/p/id/4683 -> 6534 doesn't have timers
http://forums.ni.com/ni/board/message?board.id=40&message.id=5167 -> Just a phase probleme
http://forums.ni.com/ni/board/message?board.id=170&message.id=261289 -> Another board
http://forums.ni.com/ni/board/message?board.id=40&message.id=5090 -> Another probleme
I contacted NI support, they send me an exemple on "patterns" but it generates frequencies that are multiple of one with no phase consideration or duty cycle.
Any idea ?
Thanks in advance for your advice and help.
Eric.
Message Edité par Eric31 le 08-10-2007 06:52 AM
Message Edité par Eric31 le 08-10-2007 06:55 AM
Attachments:
Sans titre.PNG ‏8 KB

At best, that's gonna be a pain in the neck with that board. If you need very precise timing for freqs, duty cycles and offsets, it gets worse. If they need to change on-the-fly, it'll be near-impossible.
1. You can only operate 1 hw-timed digital output task at a time. So all 6 signals must be part of that 1 task.
2. Thus, you need to define an output buffer representing the states of those 6 bits throughout your entire generation.
3. There's a bunch of greatest common factor and least common multiple stuff needed to figure out the right output rate and buffer size, assuming you'll want to regenerate repetitively. You first need the gcf of all the transition times to come up with the right output rate since all the transitions need to occur on some integer multiple of the output period.   You'll then need the lcm of the 6 signals' periods, combined with the output rate, to size up a buffer that can be regenerated.
4. Forming the output data array may be quite complex.   Personally, I always do my stuff with raw arrays and I know that interleaving all those signals can be quite tricky, especially if there may be an occasional simultaneous transition.   However, I suppose there's a chance that some of the digital waveform functions *might* make this step a bit simpler. I just haven't played around with that format much.
Editorial opinion: Buy a 6602. You can trigger 6 counters to start with different initial phasing, and will be able to change freq & duty cycle on-the-fly for each one independently.
Editorial opinion #2: For higher channel count apps, I'd also recommend checking out the DIO-64 board put out by Viewpoint Systems, an NI Alliance partner. It gives you much smarter hardware-timed DIO. With the NI board, there can be huge amounts of redundant info in the DO buffer. If you find that your GCF is 10 microsec, but the time between consecutive transitions is 10 millisec, you'll have to stuff 1000 identical values into your output buffer.   With Viewpoint's board, your data acq buffer *only* contains data at the transition times. Timing is controlled by delivering a separate array defining the timestamps that correspond to those transitions.
-Kevin P.

Difficulty synchronizing data from VISA resource with data from a physical channel

Essentially what I'd like my program to do is control the electrical power going (sourcing either current or voltage) into a light and measure the intensity of the light at a given power level, and then do this automatically for ~1000 increments of the source voltage/current.
One of my lab partners made a program a while back which does what we want("LIV Sweep Rev.vi" the first image in the link at the end of this post), but it's a bit sloppy: the program is able to interact with the power supply (connected directly to the computer via USB) and make it turn on, increase the voltage/current, and record the "IV" characteristics just fine. The program can also interact with the light detector (connected via NI-BNC 2110) and gather the photocurrent data. The problem, however, is that the data wouldn't be in sync. The photocurrent data for when the light was actually supplied with 1V would be improperly recorded in the cell for when 2V are applied to the light. To fix this problem my lab partner added in a time delay so that the detector will pause for ~0.35 sec (user-specified in front panel) before gathering data.
The program works, but I figured there had to be a better way. The thing about the timing, however, is that it changes from run to run. Sometimes a delay of 0.45 s works well, and other times the power supply will have shut down while the detector is still gathering data (and thus we miss the low end of the sweep). Other times the detector will turn on too early, which will cut off the high end of the sweep.
(Note: I have next to zero experience with LabVIEW, but I know a little bit of java, so I understand most programming jargon)
I spent all day yesterday trying to find out how to synchronize two data aquisitions (my attempt is shown in the "LIV SweepSummerDuncanRev3.vi" in the link at the bottom, 2nd image) but I'm running into trouble when I try to trigger the sample clock using the VISA Resource Name for the power supply.
The programs can be viewed here:
http://imgur.com/a/Up3eS
I'd really appreciate any and all advice that you folks have to offer!

Change the code such that rather than using a ramp from your power supply, just output a single value. Then do your measurement and then step to the next value in your ramp.
You can use a State Machine (SEarch if you do not know that term in LV).
Some gear will allow specifying a ramp driven by an external clock. If you widget can handle an external clock/trigger that approach could run faster.
Without hardware to syncronize the various sub-systems you would have to resort to using a RT OS and depending on the instruments, even that could be hit-n-miss.
So re-code to step-measure-step-measure etc.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

Fluke 284 Arbitrary Waveform Generator Phase Lock Channel Synchronization

I am using Fluke 284 Arbitrary Waveform generator (not NI instrument). I am trying to make use of LabVIEW to assign channel 1 as the master and the other three as slaves to phase lock the slaves and change precisely the phase offset with respect to the master.
The error explanation that I received was:
Possible reason(s):
Error -1074000000 occurred at Fluke 28x Series.lvlib:Error Query.vi
Instrument reports:Inter channel lock not possible. Refer to manual for more details.
Complete call chain:
Fluke 28x Series.lvlib:Error Query.vi
Fluke 28x Series.lvlib:Configure
I would like to know of there is any other block that I need to add to control the output of the master and the input of the slave. I have referred to chapter 13 of the users manual which I have also attached.
Attachments:
Fluke 284 Manual.pdf ‏1890 KB
Fluke_Block_Diagram.png ‏130 KB

Don't know what the problem is, but the error number is strange. It's not a standard error for Windows, LV or the instrument in question...
Mike...
Certified Professional Instructor
Certified LabVIEW Architect
LabVIEW Champion
"... after all, He's not a tame lion..."
Be thinking ahead and mark your dance card for NI Week 2015 now: TS 6139 - Object Oriented First Steps

Synchronization of AI channels on PCIe6351 card?

Hi everyone,
I have a quick question; hopefully an easy one to answer...
I am acquiring multiple AI signals from the card above... When sampling these AI signals, is it necessary to include a "trigger" in the DAQmx code necessary for data synchronisation, or is setting the sample clock source to the "PCIe6351/ai/sample clock" sufficient?
Thanks,
Jack
Solved!
Go to Solution.

Hi Dennis,
Thanks for the reply... So I can e assured that all AI signals acquired in the code below are synched?
Also, for clarification, what is the "On Board Clock"? Does it refer to the clock on the PC or the DAQ card... How does tjhis differ form the ai sample clock?
Thanks,
Jack

Hi!!I have some problems with the synchronization of my system, and I don´t know how I can solve this!

My system consists of differents vi´s.Some of them are used to read analogic inputs and digital inputs, using the serial port.
The values that I read, go through differents vi to be proccesed, and at the end, these values are written in a file, all the values in the same file.
My problems of synchronization are:
1. I have one vi per input, and all of them use the same serial port. How can I synchronize them to avoid serial port conflicts? (using queues??) And how can I do, to get differents number of samples/sg with every vi? I have to introduce them in differents loops?
2. The other problem is that, for example, I want to read 8 samples/ s but I only want
to write two samples every second. How can I do to avoid stoping the data flow?
Maybe, my problem is that I want to see the system like
one block, and I am ignoring that, although all the system works like a block, every part works using a different "speed". I don´t know how tools can I use to combine two ideas, working like an only system( data dependence, some vis need the data of others vi to be able to work) and working getting differents numbers of samples/s.
I have seen that there are synchronization vis. I am going to glance at them, maybe I find something interesting but I amsure that your help will be very useful?
Thanks in advanced!!
Graci

I think you are pretty much on the right track with Queues. I have implemented a very similar system myself using queues. I have even different systems connected to the serial interface, but I can synchronise them all easily using the following method:
1) Make a VI for opening the serial port and reading the data. The data (parsed to extract the numbers required) are then stored in an array (Shift register). If the pre-determined time delay is reached (say 5 seconds) the values are averaged and written to a Queue with a defined data type. Save the VI as a VI template.
2) Make a VI which starts an instance of this template VI for every channel you require. You can use VI Server to set values on the new front panel before you set it's property to "running". Setting a timer value for all can help ensure that the X-Axis of your timing values (if you require) are synchronised. The clue here is to define and initiate a QUEUE per channel. (RS-232_1, RS-232_2 and so on). The QUEUE name needs to be passed to the VI Template before running so that each VI running in the background has an individual link to the main program.
The second part of this VI simply uses an array of QUEUES, polling them one after another with a timeout of, say 5 ms. If data is found, append accordingly to an XY-Graph.
I also add in some "Auto termination" in the VI template (MAx number of values allowed in the QUEUE for example) to allow for the background processes to find out if the front-VI is still running or not. There are many different ways to do this, shoose the best for your application.
With this approach, it's not even important what instrument is connected as long as data is written in a specified format into a known QUEUE. I have also developed systems where I can transparently (almost) drop in VI Templates for instruments with similar functionality (output data) from different suppliers.
I would post an example, but the time it would take me to clean it up is too large. Sorry. If you have any further questions, don't hesitate to ask further questions.
Shane.
Using LV 6.1 and 8.2.1 on W2k (SP4) and WXP (SP2)

How can I get information of the switched channel in an example program from NI?

When I use the example program "niSwitch DMM Switch Synchronous Scanning.vi", I'am not able to get any information, wich channel is active. Wich block has to be used and how must I connect it to see this.
The other one is, that te sytax for the scan list, described in the help does not switch through the channel, how has this to be modified?
Im using a SCXI 1128 in Slot 8 of a 1052 combo chassis and the DMM is a PXI 4070 in the Slot 2.
Attachments:
niSwitch DMM Switch Synchronous Scanning.vi ‏145 KB

Bigmatzeman,
finally, i got a working solution which should suite you!
Ok, here is what you have to do/what is missing in your setup:
First, you can leave the modules as they are already inserted in your chassis.
Second, you will need a special AUX-cable to synchronize the DMM and the switch. You can find infos on that here
in the section "Single Module Scanning". Connect the cable between the
DMM and the front connector of the switch (1331, scanadv. and
externaltrig.)
Third, i advise you to use the HV8-BAN4-cable to connected the signals
to your DMM. This is because the cable is specified for the appropriate
voltage the DMM and the switch are capable of (300 V). The connection
of the HV8 can be done in the 1052 above the PXI-slots.
And at last, open the example "niSwitch DMM Switch Handshaking.vi", set
the devices to the appropriate identifiers. Select for both triggers
for the DMM "External" and for both triggers of the switch "Front
Connector". Change the Scanlist to "ch0:5->com0;". If you are using
the HV8-BAN4 instead of direct front connector-connection, you have to
change the code a bit, too: insert a "niSwitch Connect Channels" VI in
front of the "niSwitch Configure Scan Trigger" which connects "ab0" and
"com0".
Ok, now the example should work correctly and you should be able to use all other SCXI modules together with your 6281.
Hope this helps,
Norbert
[Edit]: My setup was 4072, 6052E, 1127 and 1011 (instead of 4070, 6281,
1128 and 1052 as you do). But that should work just the same way.
Message Edited by Norbert B on 08-04-2006 10:36 AM
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

How to Use a Certificate for Two Way SSL and another certificate for WS Security Header at Client Console Application(C# Dotnet)

Hi,
I want to consume a Java Web service from Dotnet based client Application. The service require one Certificate("abc.PFX") for Two Way SSL purpose and another certificate("xyz.pfx") for WS security purpose to be passed from client Application(Dotnet
Console based). I tried configuring the App.config of Client application to pass both the certs but getting Error says:
Could not establish secure channel for SSL/TLS with authority "******aaaa.com"
Please suggest how to pass both the certs from client Application..

Hi,
This problem can be due to an Untrusted certificate. So you need just full permissions to certificates.
And for more information, you could refer to:
http://contractnamespace.blogspot.jp/2014/12/could-not-create-secure-channel-fix.html
Regards

What is the recommended way to do multiple channel, single point sampling for control with an NI PCI-6255 in RLP?

Hello,
I am writing a driver for the M-series NI PCI-6255 for QNX. I have downloaded the MHDDK and have all the examples working. I have also enhanced the examples to do interrupt handling (e.g. on AI_FIFO interrupt or DMA Ring Buffer interrupt). My ultimate goal is to write a driver that I can use for closed-loop control at 500 Hz using all 80 channels of the NI PCI-6255. I may also need to synchronize each scan with a NI PCIe-7841R card for which I've already written a driver. I want an interrupt-driven solution (be it programmed I/O on an interrupt or DMA that generates an interrupt) so that the CPU is available to other threads while the 80 analog inputs are being read (since it takes quite a while). I also want to minimize the number of interrupts. Basically, I will need to collect one sample from all 80 channels every 2 milliseconds.
There are many different options available to do so, but what is the recommended technique for the NI PCI-6255 card? I tried using the AI FIFO interrupt without DMA, but it seems to interrupt as soon as any data is in the AI FIFO (i.e. not empty condition), rather than when all 80 channels are in the FIFO, so more interrupts are generated than necessary. I tried using DMA in Ring Buffer mode to collect a single sample of 80 channels and interrupting on the DMA Ring Buffer interrupt, which appears to work better except that this technique runs into problems if I cannot copy all the data out of the DMA buffer before the next AI scan begins (because the DMA will start overwriting the buffer as it is in ring buffer mode). If the DMA is not in ring buffer mode or I make the ring buffer larger than one 80-channel sample then I don't have a way to generate an interrupt when one sample has been acquired (which I need, because I'm doing control).
I saw something in the documentation about a DMA Continue mode in which it looks like you can switch between two different buffers (by programming the Base Count/Address with a different address than the current address) automatically and thereby double-buffer the DMA but there is no real documentation or examples on this capability. However, I think it would work better than the Ring Buffer because I could interrupt on the DMA CONT flag presumably and be copying data out of one buffer while it is filling the other buffer.
Another option would be DMA chaining, but again, I cannot find any information on these features specific to the NI DAQs.
I tried interrupting on AI STOP figuring that I could get a single interrupt for each scan, but that doesn't appear to work as expected.
I know that DAQmx on Windows has the ability to do such single sample, multiple channel tasks at a fixed rate so the hardware must support it.
Any suggestions would be appreciated.
Thanks.
Daniel Madill

Hello,
The interrupt that will happen nearest the times that you need is the AI_Start_Interrupt in the Interrupt_A group. This interrupt will occur with each sample clock. By the second time this interrupt fires, the AI FIFO should have the samples from the first conversion. If it is easier to use programmed IO, you can read the samples out of the FIFO until you get all 80.
Additionally, you can set the DMA to send samples as soon as the FIFO is no longer empty...instead of waiting for half full or full. This change will reduce latency for your control loop. You can set AI_FIFO_Mode in AI_Mode_3_Register to 0. By the second time this interrupt fires, you should be able to check how much data is in the DMA ring buffer and read the 80 samples when they are available. You can make the ring buffer larger than 80 samples if you see data getting overwritten.
There is no interrupt associated with 80 samples being available in the FIFO or 80 samples being available/transferred by DMA to the host. X Series has much more flexibility with these interrupts.
I hope this helps!
Steven T.

Feature Request: Modify Audio Channels from the Timeline in Premiere CC.1

Why have we never been able to modify audio channels once a clip is added to a sequence? I'm guessing there's a good reason Adobe has never had this feature. The lack of this feature is explicitly mentioned a few times in the audio section of the Premiere Pro CC manual, as if it's a benefit NOT to have it. But the benefit is entirely lost on me.
In my workflow, I synchronize my double-system video and audio before anything else. I do this in PluralEyes (and unfortunately I'll have to keep doing it this way, until PPr's sync on sound feature gets fixed—for me it's never worked).
Once PluralEyes has synched hours of footage and audio and placed it on a PLURALEYES GENERATED sequence, I import that sequence into Premiere. Note: Since I haven't gone through the step of creating a blank New Sequence, I don't get the opportunity to setup my audio channels the way I'd like in my sequence (i.e. Stereo recordings should occupy ONE track, not two. Typically these tracks come from a camcorder and are used for reference sound, or ambient noise at best, so I don't need them taking up precious real estate in my sequence).
Since PluralEyes doesn't merge my clips for me (or rather I don't want it to since I'm usually doing multi-cam sequences where I want to keep all 14, or so, tracks of audio), I have to go through the clips one by one and do a Merge Clip action once I've adjusted any minor sync issues, or determined which tracks I actually want to keep. But, and here's the key, at this stage I'm not yet 100% certain whether I'll use the nat sound or the wireless mic sound (separate audio recorder, etc). Furthermore, I don't really want to think about that at this point. After I've spent a few mind-numbing hours just ingesting files, sync and merges, I want to jump into the edit as soon as possible, before I lose all will to finish this edit. If I modify my audio channels right now, I'll lose the ability to make some creative decisions later on.
I'd like to start whacking out an assembly edit. And then after I have a rough program, I can start looking at which audio tracks (nat or double-system) I like better in each circumstance. This is where I become stumped that at this stage, I'm locked into whatever audio channels I've setup (or not) in the earliest stage of the edit, back before any media was actually on the timeline! It's at this stage, and usually only at this stage that I want to start Modifying Channels—FROM THE TIMELINE!
Does anyone else feel the same? If not, what is your workflow?
Feature Summary: Right-click on any clip, (master clip, sub-clip, merged, nested or otherwise) in the timeline and choose Modify > Audio Channels.
I've submitted a feature request, with a link back to this discussion, so please pipe in—especially if you want this feature.

Maybe its your specific workflow, asset mangement and use of a 3rd party application that runs you into an issue.
Take Pluraleyes out of the equation, I would imagine my workflow is the same as most everyone else. Find footage in the Media Browser. Import it. Create a Sequence and start editing. Use the Source Monitor to set Ins and Outs and then Insert the clip on the timeline.
At this point, if the audio needs to be modified, it cannot be done on the Timeline. The workaround, of course, is to Find Clip in Project; Modify Audio in the Project panel; then do a Matchframe to replace the audio clip. This works, but requires many steps per clip. Also, Matchframe in Premiere Pro CC is not reliable.
Introducing a new Modify Audio (from Timeline) feature as seen in the fantasy screenshot above would solve it.
I dont use Pluraleyse ...which I understand to be a synching aid...so I dont know the workflow it imposes on one .
Returning to Pluraleyes workflow, this issue is compounded by the need to Modify almost ALL audio. Every stereo track is separated and placed on two MONO tracks taking up precious real estate. This is not something Pluraleyes imposes. It is due to the way Premiere imports XML timelines. This would apply to sequences imported from Final Cut as well.
In this case it would be ideal to select both the LEFT and RIGHT mono tracks and choose "Modify > Audio Channels > Merge L/R Mono to a Single Stereo track." Ideal? No. I guess Ideal would be if Premiere would interpret XML files properly in the first place.
What actually is the issue with Premieres synch and merge workflow in your case?
Audio Sync in Premiere Pro doesn't work. Period.

Error while running Profile synchronization (Event ID: 6398)

Hi,
We had our User Profile Synchronization service in stopped state for quite some time due to our SQL server having "Named Instance". Since RTM version had this compatibility issue, I patched my farm with SP2 + FEB 2014 CU.
Now the "User Profile Synchronization Service" is running (I started it and now it was a success). However when I start the Profile sync job (either INC or FULL), it fails. This is the error I get:
The Execute method of job definition Microsoft.Office.Server.UserProfiles.UserProfileImportJob (ID c10650ed-0935-47a5-b3ce-a307c576ad9a) threw an exception. More information is included below.
Operation is not valid due to the current state of the object.
Log Name: Application
Source: Microsoft-SharePoint Products-SharePoint Foundation
Date: 6/25/2014 3:47:16 AM
Event ID: 6398
Task Category: Timer
Level: Critical
Keywords:
User: ST\sps02-svc
Computer: EMEA-MOSS1.st.stroot.local
Description:
The Execute method of job definition Microsoft.Office.Server.UserProfiles.UserProfileImportJob (ID c10650ed-0935-47a5-b3ce-a307c576ad9a) threw an exception. More information is included below.
Operation is not valid due to the current state of the object.
Event Xml:
<Event xmlns="http://schemas.microsoft.com/win/2004/08/events/event">
<System>
<Provider Name="Microsoft-SharePoint Products-SharePoint Foundation" Guid="{6FB7E0CD-52E7-47DD-997A-241563931FC2}" />
<EventID>6398</EventID>
<Version>14</Version>
<Level>1</Level>
<Task>12</Task>
<Opcode>0</Opcode>
<Keywords>0x4000000000000000</Keywords>
<TimeCreated SystemTime="2014-06-25T03:47:16.577398600Z" />
<EventRecordID>484089</EventRecordID>
<Correlation ActivityID="{A6F9E131-4925-4889-A759-5F873C4B2E74}" />
<Execution ProcessID="9140" ThreadID="7372" />
<Channel>Application</Channel>
<Computer>EMEA-MOSS1.st.stroot.local</Computer>
<Security UserID="S-1-5-21-1058282146-1732951074-3797079023-18290" />
</System>
<EventData>
<Data Name="string0">Microsoft.Office.Server.UserProfiles.UserProfileImportJob</Data>
<Data Name="string1">c10650ed-0935-47a5-b3ce-a307c576ad9a</Data>
<Data Name="string2">Operation is not valid due to the current state of the object.</Data>
</EventData>
</Event>
Here are the observations:
- User Profile Synchronization Service - Running
- User Profile Service - Running
- Both FIMS - Running
- On User Profile Service Application page, I see "User Profile Sync is not currently provisioned."
- If I go in for "Configure Sync Connections" I see --- The query returns nothing.
- On clicking "Create new connection" I get this error:
What could be wrong now?

Clear the file system cache on all servers in the server farm on which the Windows SharePoint Services Timer service is running. Please refer to the following link:
http://ahmedmadany.wordpress.com/2011/05/08/user-profile-service-an-update-conflict-has-occurred-and-you-must-re-try-this-action/
Make sure that the SharePoint Farm account, which is created during the SharePoint Farm setup, is a member of the local Administrators group where the User Profile Synchronization service is deployed.
Make sure that the SharePoint Farm account is able to log on locally on the server where User Profile Synchronization is deployed.
More information are provided in the link below:
http://www.sysadminsblog.com/microsoft/user-profile-service-an-update-conflict-has-occurred-and-you-must-re-try-this-action/
http://www.sysadminsblog.com/microsoft/event-6398-microsoft-sharepoint-administration-spsqmtimerjobdefinition-exception/
If this helped you resolve your issue, please mark it Answered

Synchronize AO channels in dotnet

Similar Messages

Maybe you are looking for