Effects of using differing sample rates?

I recently had a great deal of trouble rendering a 10 minute sequence. I thought it was because I had some effects (pan and scan stills layered on top of each other).
After much time and hair pulling, I finally was able to render the sequence and export it to DVDSP.
However, it turned out that the DVD had skipping issues.
I went back to look at my sequence and noticed that while my video was all in 48K, the narration track was at 44.1. Could this difference be the reason I'm having so much trouble with this sequence? Should I convert the narration to 48K? I haven't gotten any error messages regarding the audio or sample rates and it seems the audio has been rendering ok.
I'm using FCP3.
Thanks for any advice.

I think it's likely that the 44.1 is throwing it off. You can take the narration clips from your timeline directly into a bin, export and reimport as 48kHz, and replace them on the timeline. Be aware of whether they still fit in their slots or not, and see what happens.
If that was it then you get to give yourself points!

Similar Messages

Wave form chart for different sampling rate

Hi
All,
I have to use different sampling rate to get pressure data. Can I use the waveform chart to monitor the pressure data with time?
If not, what kind of graph should I use?

Different sampling rate than what? A chart or graph can be used. Depends mostly on how you want to update it.

How can I add a curve with a different sample rate behind another curve to show it like one in the report

I saved two curves with different sample rates with signal express in waveform.
Now I want to add the curves behind and show them in a report.

Hello MReizner,
Both the DIAdem VIEW and REPORT panel use the time information from your Waveform channels (make sure they actually have the waveform symbol, not the numeric data channel symbol in the Data Portal) to plot the data in the same axis system.
In the example below I have two waveforms, one sampled at 5 Hz and one sampled at 1 Hz, both in the same axis with the same time channel. All I did was drag the data from the Data Portal onto the axis. DIAdem automatically takes care of creating the correct time channel and plotting the data with the correct points if the data is stored as a waveform.
I hope this answers your question, please let us know if further clarification is required ...
Otmar D. Foehner
Business Development Manager
DIAdem and Test Data Management
National Instruments
Austin, TX - USA
"For an optimist the glass is half full, for a pessimist it's half empty, and for an engineer is twice bigger than necessary."

Different Sampling rates for different channels in Analog Input

Hi,
I would like to acquire data at different sampling rates on different channels say ACH 0, ACH 1 and so on. I have a PCI 6052E board and NI DAQ 6.9.2. Also is it possible to simultaneously perform Analog output on two different channels along with the Analog input? What will be the problems/consequences as far as the system resources are concerned. I am a beginner in this area and would greatly appreciate any help/pointers for my queries.
Thanking you in advance
Deepak

Search the eaxamples that ship with LV.
Theer is one called simultaneous input and output or something like that.
It will get you started.
re: multiple scan rates. This is acoomplished by sampling all channels at the highest rate and throw away the expttra samples you do not need.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

Parallel acquire 2 signals with different sampling rate on 2 cards

Hi NI,
I have cDAQ-9178 and NI 9221, where sample rate is 10kHz and NI 9219, where I need sample rate about 10Hz, it's possible this confiruration for parallel acquire?
Thank you.
Neolker

Hi Neolker,
Do the tasks need to be synchronized or are totally independent tasks?
If the tasks are independent, than you basically have to create two different tasks, even in two different loops that will run with different sampling rate.
You can have them in the same loop the reading if you assure that the data are transfered in chunks to application memory according to tha sampling rate (Ex. for 10kS/s rate you can transfer data with 1kS chunks and for 10S/s rate with 1S chunk).
If you want to synchronize them, you will need a counter that will divde the sample clock from 9221 and route it to 9219.
Let me know if you need more help.
Best regards,
IR

How to acquire data from 2 chs of the same DAQ card at different sampling rate

I am using single DAQ card (either 6013 or 6014) in my system i want to acquire data from 2 (or more) channels with following requirements
1. sampling rate of each channel should be independant of each other (say one is 20 Hz and other is 15 kHz)
2. data from all the channels should be acquired simultaneously.
3. coding must be done using DAQmx VIs
I have tried out following things
1. I created separate task for each channel: i found out that two tasks can not run simultaneously even though the channels are different
2. I tried out single task with two channels included in it. and i used 'channels to Read' property to determine from which ch. i want to acquire data: this method works fine if the sampling rates are same. but if i change the sampling rate of one channel it gets reflected in other channels as well.
can somebody help me out to solve this problem.
i will appreciate if somebody can post the sample code as my deadline is approaching
Tushar Jambhekar
[email protected]
Jambhekar Automation Solutions
LabVIEW Consultancy, LabVIEW Training
Rent a LabVIEW Developer, My Blog

Hi Dennis Knutson
Thanks for your suggestion.
Tushar Jambhekar
[email protected]
Jambhekar Automation Solutions
LabVIEW Consultancy, LabVIEW Training
Rent a LabVIEW Developer, My Blog

Acquire 2 signals with different sampling rate?

Hello,
i am using first time labview signal express together with a NI pci 6251 & NI Scc 68 device.
There are 2 signals I wish to acquire over a 2 week time period. They should be saved in several tdms files. One is a dc Voltage.(sampling rate 100Hz) The second is voltage given by an acceleration Sensor. (sampling rate 4kHz)
When I try to acquire them I couldn't figure out how to set the sampling rate for each signal, only for all signals.
When I created two DAQmx assistant acquire tabs. An error occurs always: "the specified resource is reserved".
So my question is is it possible to do this?

hello markus_umd,
no, it is not possible to create different samplingrates for individual channels on one daq device.
if you want to record one channel slower, you must reduce the samples for that channel in your program.
if the channels have a common divisor, you could implement your task. you find more information here:
Sampling Different Channels at Different Rates with NI-DAQmx
kind regards,
robert h
NI germany

Audition 3 seeing a different sample rate setting than what the device shows

Hi,
I have just installed Adobe Audition 3, along with the 3.01 patch, on a brand new system running Windows 7 64 bit. The mother board is an Asus Sabertooth X58 using Realtek High Definition Audio. The device drivers show that the audio sampling rate for line input is set to 24 bit 192K. I wanted to set it to the maximum that the sound card would allow to test performance and audio quality.
The problem is when I bring up Audition 3 and hit record, I get the message "We do not support recording when your file does not match your hardware sample rate. Your current hardware sample rate is 44100Hz". Clearly this is not the case since the Line In Properties - Advanced tab is displaying "2 channel, 24 bit, 192000 Hz (Studio Quality).
Under Audition's Audio Hardware Setup it shows only one choice for Audio Driver: Audition 3.0 Windows Sound. It also displays Sample Rate: 44100Hz, Clock Source: Internal, Buffer Size: 2048 samples with no way to change these values.
If I click on the Control Panel button I get:
DirectSound Input Ports:
Device Name: Line In (High Definition Audio Device
Audio Channels: 2
Bits per Sample: 16
Anyone know of how I can change these settings to get Audition to agree with the device settings?
Thanks
Dale

DaleChamberlain wrote: Anyone know of how I can change these settings to get Audition to agree with the device settings?
I'm afraid that life is nowhere near that simple. The main issue here is that Audition, in common with most audio software, uses a driver system called ASIO to talk to the sound device - this cuts out a lot of the OS and reduces the latency of the system considerably. There are several problems with ASIO though - the first being that it only supports a single device per system (or sometimes multiple identical devices if the manufacturer can make them look like a single device), and with software designed to use this driver, then to use any other driver (like a native Windows one) you have to use a converter stage like ASIO4ALL. This will convert the ASIO streams to WDM, and let you use multiple sound devices - but with increased latency.
It's the second problem that's really going to stuff you though - and that is that quite reasonably, ASIO is limited by its inventors to run only at three sample rates; 44.1k, 48k and 96k. So there's no way you can run at what you think might be a higher quality setting. All settings above even 48k are making your sound device work much harder, and for what? All that happens is that you increase the potential frequency response to way beyond the human hearing range - to no purpose at all. You don't have sources that can produce useful output at these frequencies, and you certainly don't have the means to reproduce them. This has all been well documented and explained before, so I'm not going over all that again. In a nutshell, Nyquist points out that any digital sampling device has a frequency response limited to a maximum of half of the sample rate, so for 48k that gives us a frequency response up to 24kHz - comfortably higher than any adult can hear by quite a long way. Anything you sample and record beyond this by using even 96k is nothing but noise as far as humans are concerned, and unpercievable noise at that.
So what the line input properties tab is saying is, if you have a non-ASIO driver designed to support all potential rates, possible. You don't have an ASIO driver available, because it's a built-in sound device, and anyway you've already pointed out that it's using the Audition Windows driver (a cut-down version of ASIO4ALL, effectively), so a conversion is already taking place. What Realtek refer to as 'High Definition Audio' is no such thing - all on-board sound devices of this nature are of universally low quality, and to improve this you'd need an external device - of which there are many available, usually with dedicated ASIO drivers. But none of them will work with ASIO beyond 96k, simply because the standard doesn't support any higher rates.
If you download and use ASIO4ALL (it's free), then you will get an additional control panel which will show you exactly what your sound device is capable of doing as far as Audition or any other ASIO software is concerned, and this is a useful diagnostic tool anyway, so it's worth doing. You just select this option when installed, instead of the Audition Windows Driver.
I'm sorry to be the bearer of what seems like bad news, but actually, it isn't. You will percieve no quality difference at all running at anything beyond 48k sample rates; all you will be doing is wasting your computer's resources unnecessarily. You waste both processing resources and hard drive space by processing at ridiculously high sample rates, and there are zero returns.

OMF - Mixing Different Sample Rates

Hello -
So, I've got an OMF file from a FCP project that I opened in Logic and have been mixing without any issues. Interestingly, all of the audio files associated with this project are 44.1 kHz, but I am mixing in 48k. But everything is right (sounds right, looks right, syncs with video correctly).
But... if I open the same files in an external editor or quicktime, they play back incorrectly. And if I save a file in a different program, even if it is still at 44.1, and bring it back into logic, it plays back incorrectly. This is problematic if I need to edit an audio file somewhere other than within logic (say I want to do some noise reduction in soundtrack pro).
Anyone run into this issue or have any ideas about how this happened?

There are a few ways to look at this.
1) Regions in the arrange all play back at the session sample rate. Example: 44k session, 96k audio file in arrange=slower playback
2) Logic automatically converts output sample rate so you can record independent of CA devices. Example: You have a session recorded at 96k, your interface is not connected, Logic will load Built in Audio, Logic runs tyhe sessions at 96k and converts the SRate of the session to match the supported sample rate. So you can run sessions at unsupported sample rates, this rarely makes sense if you cannot capture your audio at session sample rate (if needed).
3) There are a few other options for handling this, such as EXS24, which automatically handles SRC.
4) When Importing Audio Files there is a song preference which you can en/disable to automatically convert SR upon import.
I think your friend may have referred to point #2 and it was interpreted as point #1...perhaps. Hope this clears things up. J

Different sample rates from different cameras

I took some video with my Panasonic video camera which samples at 48kHz. At the same time I was shooting with my Digital camera which samples at 8kHz. I am trying to pan between the two of them but the timing does not match up through the video. The clips are approximately 3 min and 20 seconds. Any idea how I can get these two things to synch up?
thanks

Video camera is Panasonic. PV-GS250
Digital camera is Lumix DMC-LX2
The Browser in FCE shows me that the video camera's audio format is 16-bit integer and 48kHz, the audio from the digital camera is 8-bit integer and 8kHz. I assumed that was the bit depth and sample rate.
Good question on the video format of the digital camera. I just assumed it was DV-NTSC, and so thats how I imported it. I will check into that as it seems to be the most likely solution.
Thanks!

What's the best way to sample multiple AI's with different sampling rates under one task?

I'm using a PCI-6221 card and CVI 7.1.
I have a tri-axis vibration sensor and two other pressure transducers.
I want to take 10k samples from each vibration axis at 80 kHz. This is
possible by configuring the scan rate of the "vibration task" to 240
kHz. (The card maximum is 250 kHz).
I want to take 1k samples from each pressure transducer at 250 kHz. This happens very infrequently.
Each measurement is required by a separate task.
I thought I could do this by setting up three finite tasks (1
vibration, 2 pressure), but DAQmx won't let me run more than one AI
task at a time. I've read other posts here, and I realize I have to
add/remove physical channels on-demand.
What is the best way to optimize this setup so that I'm not hogging up system resources?
Should I do the following?
1. Stop the task
2. Remove the vibration channels from the task
3. Add in a pressure channel
4. Configure the pressure channel
5. Start the task
6. Take the pressure samples
7. Stop the task
8. Remove the pressure channel from the task
9. Add in the vibration channels
10. Configure the vibration channels
11. Start the task
Also, the vibration portion is running in finite mode, but I'm looping
it. Should I switch it to continuous and run the "DAQmxReadAnalogF64"
to sample the latest 10k samples. (If the task is continuous, would I
pull the latest 10k samples, or would I pull some old buffered samples
instead?)
Thank you,
Nobody

Hello Nobody,
If you configure your task timing to acquire a finite number of
samples, then you can only read the number of samples that you
specified in your configuration. Once you try to read more
samples, you will receive the error you are seeing.
If you configure your task timing for continuous acquisition,k then you
can read samples indefinitely. Any given DAQmx Read will read the
oldest unread samples in the buffer.
If you are going to be switching between different tasks, you will definitely need to stop one before you start the other one.
I hope this helps!
Eric
DE For Life!

NIDAQmx: how to simultaneously acquire channels at different sample rates?

I have a PCI-6229 (M-series card). It has an advertised throughput of 250K/sec. I'd like to configure the card to simultaneously sample eight (8) AIN channels at 25K/sec, and none (9) AIN channels at 5K/sec. This results in a usage of about 245K/sec out of the available 250K/sec, which seems "do-able". But how do you setup the task(s) in NIDAQmx to do accomplish this? Are there any examples (I'm using the ANSI-C API) which illustrate how to do this?
I can't just set all 17 channels to 25K (and decimate the data for the slow channels), because there is not enough bandwidth. I was also unable to setup two tasks (one for high-speed, one for low-speed) because I get an error when I start the
second task. Is there another way to accomplish this?
Thanks in advance for any help!

Greetings,
I would take a look at this KnowledgeBase that describes the process between doing this task.
http://exchange.ni.com/servlet/Redirect?id=12678799
Unfortunately, a text base example program does not currently exist, but if you have questions we'll do our best to help you out.
Regards,
Anuj D.

IPhone: AudioQueue - is it possible to change the sample rate?

I've been playing around with the AudioQueue stuff for a few days and it's all working fine.
I was trying to build a low-latency playback system by making the streaming buffers the same size as the audio file and pre-loading the buffers (which works fine) but I've hit a snag.
I've been trying to get the streaming to work at different sample rates so that I can play back the same sample at different pitches. I managed to do it by modifying the sample rate in the AudioStreamBasicDescription structure but in order to actually make the stream playback at the new rate it seems you have to create a new output, reload the audio file into the buffers and re-enqueue the output queue before starting playback again, otherwise the sample rate change has no effect.
There is a method to set queue properties; AudoQueueSetProperty() but unfortunately the sample rate Property (kAudioQueueDeviceProperty_SampleRate) is read-only
Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
Thanks,
Neil

Dan,
there is one point in your understanding, which i am not sure what you think about when talking about it: I understand E-series devices do not support this property change while the VI is running.
infact, you cannot change the sample clock rate during acquisition. but
this does not mean that you cannot change it while the VI is running.
you have only to interrupt the acquisition. since you want to acquire
continuous, this would have the same effect as stopping the vi, i asume.
so the best way to accomplish this task is to use an external clock.
this is e.g. often used for acquistion on rotating shafts. the
acquistionrate is always e.g. 24 points per revolution regardless of
the rotational speed of the shaft, except for a maximum frequency of
course.
Norbert B.
NI - Germany
Message Edited by Norbert B on 09-14-2005 04:16 AM
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

Diadem sampling rate

Hello!
We are using Diadem Version 11.0 and the hardware cDAQ with the voltage measurement hardware NI9229/9239.
I am trying to sample the voltage by at least 4khz, but I always get an error message that says that this rate isn't supplied by the hardware (the handbook says different things). I also changed in the options from windows timing to software timing or automatic timing, and also from hardware to software timing. Didn't work out. How can I maybe change the resolution to get a higher sampling rate?
Please help me , thank you very much in advance

Hi Lorenzo!
The problem is the NI 9229/9239 cannot sample at 4KS/s. If you check the NI 9229/9239 data sheet , it is said in the fist page you must use a sampling rate (Fs)= (50KS/s) / n; where n=1, 2, 3... 31; so it is not posible to sample at 4KS/s.
You can use sample rate (Fs)= 4,166 KS/s (using n=12); or you can use sample rate (Fs)= 3,846 KS/s (using n=13) but not Fs=4kS/s

[Q] How to set the sampling rate separately?

Hello,
I'm using LV5.1 in Windows98 with SCXI-1200.
I want to set different sampling rate in each input channel.
I've ever used "AI acquire waveforms.vi" only.
Anybody can help me?
Example codes are highly appreciated.
Regards,
Hyun-ho Lee
[email protected]

The inputs of the SCXI-1200 are multiplexed to a single Analog-to-Digital converter. Because of this, the only difference in sampling rates that are achievable would be integer divisions of a common high frequency. This is functionally identical to acquiring all channels at the highest rate, and decimating (throwing away) data from channels that need lower data rates.

Effects of using differing sample rates?

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