Adjusting the sample rate

How do I adjust the default sample rate setting to 44100?

Blastbot wrote:
If you are not going to answer this because it is on an older product, that is lame. I tried to get on a forum for said older product, but that is grayed out from the list of products you have a forum on, which is also lame.
Well, I certainly don't wish to be described as "lame," Blastbot, as it is a personal fear of mine. The forum for previous versions of Audition is available at http://forums.adobe.com/community/audition/audition_previous but since we're here and my earlier obligations today have wrapped freeing up a few minutes of my time, I'll be happy to walk you through the process for Audition 3.0.
In Audition, click Edit > Audio Hardware Setup... In the dialog that pops up, you'll see the current hardware sample rate listed and a Control Panel button that will display the device configuration panel for the selected driver. You should be able to change your sample rate by clicking that button, depending on which audio driver is selected. The default "Audition Windows Sound" driver, which is generic and best suited for audio devices without their own drivers, defaults to the Windows sample rate. If you have a good audio device with its own, native ASIO driver, you can usually modify the sample rate within their control panel.
Since you're apparently running Windows, although I'm not certain which version you have, you may need to change the sample rate of your audio device at the Operating System level. You can usually reach this menu by opening the Windows Control Panel and launching "Sounds and Audio Devices" (on Windows XP) or "Sound" (on Windows Vista and Windows 7.) Here, you'll need to navigate to your playback device selection, open the device properties, and adjust the default sample rate that you desire.
More detailed information can be found in the Adobe Audition Help documents either from the Help menu item, or the Help button located in the bottom-right corner of the Audio Hardware Setup window. Additional information about Sample Rates can be found on Wikipedia at http://en.wikipedia.org/wiki/Sample_rate
I sincerely hope this helps answer your question and relieves me the burden of lameness, at least as far as this issue extends.

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ADjusting the sampling rate on a FIR filter?

How do I adjust the sampling rate on a digital FIR filter? Thanks in advance.
-David

You should really start a new thread instead of posting to one that is 5 years old.
To answer your question, it depends on your data. I don't use the DFD but with the filter functions in LabVIEW, if you pass a waveform data type to the function, then the waveform data type contains a dt value. So, set the DAQmx Read to return waveform data. If you are using low level filter functions where the input is a 1D DBL array, then the filter has to be configured. With the low level functions in LabVIEW, you use the various coefficients functions that have a sampling frequency input.

Sampling Rate- How do I check and adjust the sampling rate on my Laptop?

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The first things we need to know in order to be able to help are what are your operating system and audio device?
Also the FAQ Setting the Sample Rate in Windows Vista and 7 may help.

How can I change the sample rate (delta X)?

Hi, I create vi which can measure resistance, it will record the data and write to a file.
however, I want to change the sample rate which is delta x. is that anyway I can do?
I have try to put a clock inside the loop, it does slow down the speed but not the sampling rate.
Please help me!!
Thank you!

Hello Shan405,
Adjusting the Reading Rate of a DMM
As noted in the above DeveloperZone article, there are a lot of factors contributing to a reading rate of a DMM. 'Sampling rate' can only be configured in digitizer mode, which can only be achieved on the flex DMMs
Using the NI PXI-407x DMM as an Isolated Digitizer
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Is the sample rate half with the snd read waveform.vi?

Hi Guys,
I plan to use the PC soundcard for data acquiring with the standard LV "snd read waveform" vi. During programming I found that the sample rate is the half of the adjusted value. For example when I set
- the format to 8bit mono
- the buffer size to 11025 byte
- and the sample rate to 11025
then the operation takes 2000ms. I would expect 1sec.
Used system: LV7.0, WinXP SP2.
Does anybody know the reason? I have to use 44100Hz sample rate (44100 "individual" sample per secundum) in my application.
Thanks,
Tomi
Attachments:
Panel.jpg ‏18 KB
Block.jpg ‏15 KB

To be more precise, that's the time take for the below mentioned finite cycle
SI config>> SI start>> SI read >> SI clear.
Its this 'entire' process which takes 1997 milli Sec
Now, what you really have to check is wheather your 'SI read' function is living up to the sampling rate you have set!
Check that by putting your 'time elapsed' check for SI Read function alone!
Hope this helps
Regards
Dev

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.
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Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
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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.

The sample rate in my code is too low. Is there a way to make it run faster?

Dear Sir,
    I am a LabVIEW customer, and just have a question about LabVIEW.
    We are running a code of "Wirebonder Project", but find the sample rate is too low. I'm wondering if there is a way to make it run faster. (I have
attached the code here.)
   Thank you very much!
Yours Sincerely,
Xian Zhang
Attachments:
Wirebonder_Project_Code.zip ‏169 KB

Hi Xian,
What system are you running this code on and are you using the LabVIEW Real-Time Module? There is a presentation online on Optimizing and Benchmarking Real-Time code with useful related links as well as other articles in the DevZone on things like Benchmarking an Application Using LabVIW Real-Time and DAQmx.
As you can see, this is not a simple question, so take a look at the information in these links and see if you can narrow down what in your code is running slowly and how much you might be able to improve the speed.
Regards,Stephen S.
National Instruments
Applications Engineering

Setting the sampling rate in SignalExpress

I am using a cDAQ-9172 with a strain gauge module and a thermocouple module and using SignalExpress. I want to acquire data at a pretty low frequency rate (2Hz), but I am unable to use 1 sample on demand in the acquisition setup. I get the following error.
Error -201087 occurred at DAQ Assistant
Possible Reason(s):
Measurements: Task contains physical channels on one or more devices that require you to specify the Sample Clock Rate.
Specify a Sample Clock Rate.
Device: cDAQ1
I am unable to setup a sample clock rate as it is 'blacked' out by the software in the 1 sample on demand mode.
When I try to use continuous or 'N' sample mode, the cDaq is sampling at a rate of around 1600 Hz, even though I am putting in a value of 1Hz on the seti[ screen. The large amount of data will prevent me from downloading the data to Excel for further reporting. I obviouisly don't need to sample my TC module at that rate either.
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Jay

Hi Jay
I am assuming that you are using the NI-9237. This strain module has only a specific set of sampling rates. If you specify a rate that is not supported it will corers the rate to the next higher sampling rate. Also since you are using the cDAQ chassis, there is only sample clock so it must choose the highest sampling rate. So it is not possible to sample at a lower rate. Please see the following link.
If you have access to LabVIEW then I would suggest taking your readings at the fast rate and then average the samples before you write them to your file. The following link has some more information on this.
If you are limited to Signal Express, this task is tricky. If you select a N-Sample acquisition, set the samples to read to greater than two, and set the post acquisition delay under the execution control tab to 1000, you acquisition will acquire the two samples at the very fast rate, but it will wait 1000mS between each acquisition. This will slow the overall sample rate.
Chris_K
National Instruments
Applications Engineer

What is the sample rate of voice memos?

I often record lectures via Evernote audio notes on my iTouch. I want to convert those audio files to text. A piece of software named, MacSpeech Scribe has caught my eye. A review states, "a minimum sampling rate of 11.025khz" is needed for the conversion.
Does anyone know the sampling rate of an iTouch audio note?
Thanks in advance.
DaverDee

ok i've figured out the physics of this question.
The Cache Read data rate is always larger than the Cache Write data rate, because the computer would have to be rendering to Cache faster than realtime for the Write rate to be higher, which would make it unnecessary to render to cache in the 1st place. So I'm really only worried about the Cache read data rate. Does adobe have a paper that tells us what the data rate is for different sequences.
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How to coerce the sampling rate??

I think I found my problem with sampling rate.
I'm using a PCI-5122 scope card, and in many of my aquisitions, I'm setting the sample rate to 40MS/s. Apparently, this is not a valid number and the scope reverts to 50 MS/s
Later when I try to calculate cycles per second based on cycles per sample, I need the actual sample rate, and 40 MS/s ain't it.
I'm trying to coerce sampling rate.
please,.

The digitizer coerces the sample rate because of how the sample clock is derrived from the Reference Clock. The following information is on page 13 of the specifications:
http://digital.ni.com/manuals.nsf/websearch/C6B059C1BDD70101862574C8005567F1
The sample clock is created by dividing down the Reference clock (internal reference clock is 100MS/s) by decimation, and it divides it by N, which is an integer between 2 and 65530.
Thus 50MS/s uses a decimation factor of 2, and 33.3MS/s is the next valid sample rate with a decimation factor of 3. So when you specify a sample rate that is not possible, the driver automatically coerces the requested sample rate up to the next valid rate. You can obtain the actual sample rate used in an acquisition using the NI-SCOPE property "Actual Sample Rate", or the LabVIEW VI "niScope Sample Rate.vi". Using this property, you can get the values you need for your calculations.
I hope this helps!
Nathan
Product Support Engineer
National Instruments

HT5848 What is the sampling rate and codec for iTunes Radio. Is it lossless encoded?

What is the sample rate and codec for iTunes Radio? Is it lossless encoded?

I have to agree with you. There are several forum discussions on bit rate being as high as 256 kbps but I don't see how it could be more than 96 kbps based on the poor sound quality I'm hearing. I'm comparing it to an internet radio station that is 128 kbps and sounds much better.
Am I missing something?

If your Lightroom 6 is crashing in the Slideshow module, check the SAMPLE RATE of your music track

After spending over 2 hours with an Adobe technician who had control over my Windows 7 machine, I've learned something that I don't think Adobe knows yet... I upgraded from LR 5.7 to LR 6 (not CC) and began having issues when I was working in the Slideshow module. When trying to add a music track, Lightroom would crash ("unexpectedly quit"). I couldn't for the life of me figure out why. I turned off the GPU since a fair amount of chatter on the forum related to that, but it didn't change anything.
So I finally waited patiently for the Adobe Chat line to finally get a tech to respond...and he was exceptionally detailed and methodical in his approach to diagnosing my issues. I could follow on my screen as he did a variety of things that indicated he understood what he was doing. Unfortunately he got to a point where he thought (as did I) that he had found the problem/solution...which he thought was related to the bit rate. He concluded that music with a Variable Bit Rate (VBR) as opposed to a Constant Bit Rate (CBR) was causing the problem. Also unfortunate was that this "discovery" was near the 2-1/2 hour mark and we did not apply the scientific method to property determine if we had narrowed down to cause/effect.
Based on my further analysis, he was close, but not right on. It isn't the BIT rate mode that is causing the crash, but rather the SAMPLE rate. Turns out that the vast majority of songs in my iTunes library have a sample rate of 44.100 kHz--and those work just fine. However, I have about 15 songs, mostly downloaded as free background music from the YouTube creation library, that have a sample rate of 48.000 kHz--every single one of those cause my LR6 to crash.
If you're having a problem with Lightroom crashing in the Slideshow module, check the SAMPLE RATE of your music track.

If you're on Windows, check out a free program called Audacity--it's fantastic. With Audacity the process of converting a 48.000 kHz to 44.100 kHz is as simply as opening, changing the sample rate, and then exporting again, as an MP3.

Changing the Sampling rate

Hi,
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Adjusting the sample rate

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