Setting Sample Rates in Audition

Hi I am trying to record in Audition and it says my sample rates don't match. I have tried and failed at this so I keep going back to sound booth. Any ideas please?

The problem may be either the settings for your soundcard or more likely the operating systems sound settings if you are using Windows. What soundcard or audio interface are you using?
See this Audition FAQ:
http://forums.adobe.com/thread/973133?tstart=0

Similar Messages

Set sampling rate and bits on Audigy 2 Va

Hi guys, I really need your help:
I dont know how to set sampling rate from 6bit 48KHz, which the audigy 2 can be set upto 24 bit 92 KHz.
Please help me
thank you

Can you check the rear output on the soundcard with a set of headphones, to check if that works? As long as the speaker selection is set correctly (5.), then it should work.
Cat

How does Core Audio set sample rate?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

Well, here is more information:
- I reinstalled QuickTime 7 and did many other things (trashed preferences,etc)
- I set the audio hardware (Pro Tools/HD Native card) to 48k
- I checked the Digi CoreAudio Manager and it said "Connected at 44.1K", even though the hardware was set to 48K
- I played the problem videos, and all are now in sync
- I check the Digi CoreAudio Manager and it still says 44.1K
- I checked the hardware and it is now set to 44.1K
I know that the videos are 48K, because I created them in Final Cut Pro X.
Does this mean that QuickTime 7 always converts videos to 44.1K, regardless of their sample rate?

Can't set sample rate 1609

Hi,
we've recently upgraded to LV 8.6 and DAQmx 8.7 and then got problem with the data aquisition that uses the DAQmx API. For example, we have a cDAQ-9172 and 9239 AI module. The device could be user configured and a typical configuration could be a continous acq, single sample in 10 Hz. After upgrading the error -200279 "Attempting to read samples that are no longer available ... has been overwritten" has come up soon after the task was started. It turned out that the property SampleClkRate is not affected by the value that is put into the DAQmx Timing.vi, unless it was set > 1612,9, if you set 10,100 or 1000 or whatever the sample rate will still be 1612,9 when you read from the timing property.
So the buffer then of course becomes overflown, but the question is why there is a minimum sample rate like this? Earlier it was fine to set it an arbitray value and the acquistion would be in that rate.
There are many solution to get around this (read faster etc.), but it strange that the behaviour of the code can change like this from a version to another...
/Henrik
Solved!
Go to Solution.

I see one flaw in your program, you have hardware timing and software timing in one loop. The loop is limited by the software wait. (I think this is on purpose for demonstration).
I have looked at the manual for the 9239 and page 18 notices that hte minimum input rate is 1613kS/s
So that is explained, the only problem is that the DAQmx timing VI does not return an error or warning when setting a too low rate.
Ton
Free Code Capture Tool! Version 2.1.3 with comments, web-upload, back-save and snippets!
Nederlandse LabVIEW user groep www.lvug.nl
My LabVIEW Ideas
LabVIEW, programming like it should be!

G530 / XP. Playing projects in 44100 Hz makes problem. Can't set sample rate of the sound cart

Hi, I have already instal Sonar on my new Lenovo...
And what i see.... In the soundCart (conexant HD smartAudio 221) no 44100 Hz!
And i can't find something that looks like place to change it.
How can i do this?
PS: XP.
Thanx

The red line indicates the audio must be rendered. This may be caused by several things but your playback settings may be set too high for your system to handle. You may have too many audio tracks for your computer to play in realtime.
Select one of the audio files in the timeline, get properties, not the audio format and sampling rate.
Now got to Sequence>Settings and determine if the settings match your audio settings.
Your online help system can guide you through adjusting your system settings, user settings and sequence settings.
bogiesan

Set sample rate to 96 kHz in iMac

If I go to Applications>Utilities>Audio Midi Support, I can set the Sample Rate to 96 kHz, but in fact the Sample Rate is back at 48 kHz next time I look.
Question: How can I get the machine to accept and stay at the new sample rate of 96 kHz?
Robert

Hi Robert, try this...
Safe Boot , (holding Shift key down at bootup), use Disk Utility from there to Repair Permissions, test if things work OK in Safe Mode.
Then move these files to the Desktop for now...
/Users/YourUserName/Library/Preferences/com.apple.finder.plist
/Users/YourUserName/Library/Preferences/com.apple.systempreferences.plist
/Users/YourUserName/Library/Preferences/com.apple.desktop.plist
/Users/YourUserName/Library/Preferences/com.apple.recentitems.plist
/Users/YourUserName/Library/Preferences/com.apple.audio.AudioMIDISetup.plist
Reboot & test.
PS. Safe boot may stay on the gray radian for a long time, let it go, it's trying to repair the Hard Drive.

Setting sample rate to 22 kHz

Hi All,
Im trying to create a 48kbits/22kHz mp3 mixdown/bounce of my project in logic express 7.2.3. Has to be 22 kHz to play in flash player on website. The bit rate is no problem as its a setting in mp3 bounce.
Problem is when i go to select from Audio > Sample Rate i dont get 22 kHz as a sample rate, only 44kHz upwards
Ive searched manual and forums but cant seem to find an answer on how to do this. Also most 3rd party conversion software ive come across doesnt allow resampling, only rate conversion. I know i can do it in another editor, eg audacity, but i would prefer to do it all in the one program.
Can anyone help me on how to set up a 22 kHz / 48 kbits mp3 bounce?
Thanks

Thanks Noeqplease,
That really is a pity. Audacity will just have to do it for now, and i might see if i can do most of my editing in that. Pity given logic expresses capabilities.
Appreciate the quick reply

Setting sample rate for sinus analog output

Hello,
I've been trying to do something very simple : using an analog output of the card PCI 6221 to produce a sinus curve of frequency 50 Hz. For this I used a Vi to create a sinus curve and the different DAQmx VIs. But I have difficulties understanding the principle of virtual channel and I think I'm doing a mistake setting the sample rate and samples number : one time for the sinus vi, second time for "DAQmx - Timing". Should I use the same values for both of these VIs ?
On my oscilloscope, with frequency=50Hz and sample rate=1kHz, I get a null signal. Then depending on both values, I get differently rated signals. For example with f=1Hz and sr=10kHz, a sinus of frequency 0,7 Hz.
Solved!
Go to Solution.
Attachments:
Sinus analog output.vi ‏32 KB

Yes, thanks for your advice. I used the structure given in the example and now it's working fine. I'm still not sure what I did wrong though.
I would have a second question now (should I create a new topic?):
I put a continuous sine wave on the analog output. As soon as this is running, (or maybe after a short delay) I want to measure a limited amount of samples on my analog input. How can I be sure, it's not going to start measuring before the output is properly set ?
I don't think a trigger would solve the problem since I'm going to vary the output Amplitude.

Setting sample rate with Alesis IO26

There has to be an obvious solution, but I can't figure it out.
How do I get Soundtrack Pro 2.0.2 to set my Alesis IO26 to 48KHz for recording? I've changed it in the project settings and in the toolbar, but the IO26 remains at 44.1 no matter what.

That did it!
(actually, the first time I tried it, it switched to 48 KHz, but immediately switched back to 44.1 when I launched SoundTrack. But when I tried a second time it seemed to 'stick'.)
Thanks!

Parallel port sample rate

Hi,
I`m desperately in need of your help.
For my school project, I`m trying to write a labview code in order to acquire air pressure data by using 8-bit analog digital converter via parallel port. Although my signal frequency very low -100 Hz-I think I experience sampling rate problem. I want to put sample rate and frequency data in my code but how can I put it in my code?I `m giving square waveform via function generator which I need to see it on my graph in Labview. But there is something wrong in my signal in waveform chart. You can see screenshot and code attached. . Also there are some noises and amplitude is not stable . How can I fix this?
Would you please have a look at to my code and tell me how can I set sample rate and frequency?
Attachments:
CAN1.VI ‏21 KB

Hi LW-s
There isn't really a way to set the sampling rate for these VIs. You will have to use software timing. The only way to do this is with 'wait' functions inside your loop. You can also implement some logic to smooth/adjust your waveforms. For a square wave the logic is pretty simple. If the input value is above a threshold store it as the max and if it is below the threshold store it as the min. (See the example below). For sine waves this is a bit more difficult and if I were you I would probably try to implement some sort of averaging.
Best Regards,
Chris J
Message Edited by ChrisJ on 08-23-2005 05:31 PM
Attachments:
example.JPG ‏86 KB

Slow Playback Adobe Audition CS5.5 - not sample rate setting error!

I'm currently using Windows 7 and Adobe Audition CS5.5 build 1815. Soundcard is RME Digi96 Pad. Ok its an old card and the drivers date from 2005. In Adobe 3.1 and CS5.5 I get slow, around half speed, playback at 44k and 96k sample rates. YES the project sample rates and sound card sample rates are set correctly, clock is set to master on the soundcard.
However, in using Cockos Reaper 3 or 4 on exactly the same computer, same soundcard, same OS, I can play 44k and 96k wav files perfectly.
This suggests to me there is an error in the way Adobe Audition 3 / CS5.5 is handling the ASIO drivers in Windows 7 with some soundcards.
If it was a driver problem, or indeed a soundcard hardware issue, then Reaper would not perform properly.
So what's going on with the ASIO drivers in Adobe Audition with Windows 7?

I have two XLR wireless mics, which are great for my video work, but I need to do audio recordings for podcasts and wanted a mic that would be able to do just audio podcasts, as well as video podcasts, live via my laptop. So while I do agree that the audio recording from my XLR mics is great, I am not happy with the audio quality of standard computer mics I've tried and the only way I can get my XLR mics into the computer directly is through a USB connector (which I also have, I'm getting the Yeti set-up for my business partner to use).
Maybe Audition just isn't the best software option for this, but since we have CS6 Premium I had hoped I could utilize software I already had. (And for the live video casts I'm looking at starting out with Google hangouts...)
With that being said, I appreciate your reply and thought I had already set-up the hardware correctly in Windows & Audition. Here is a screenshot that shows the settings I have - my question at this point is whether the master clock needs to be changed (the audio channel mapping tab shows the left and right speaker of the default Realtek speaker):
I just double checked the properties of both the speakers and mic and they are both set to 48,000/16/stereo, as is the audio file in Audition.
I'm a journalist, not an audio pro, and I fully accept that the sound device must not be properly connected. The problem is that I cannot figure out why that is the case.
Since I need to use the USB mic and I want to be able to use the power of Audition to put together podcasts, using other software for the livecasts, I would really like to sort through this. Any and all help is greatly appreciated!

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.

Why Audition CC don't show sample rate correctly?

Hi
Using Audition CC on Mac OS X with Mackie 1220, 12-channel premium analog mixer with integrated 24-bit/96kHz Onyx FireWire I/O (Quality is amazing!)
Need to record on mono a horrible old tape recording with voice only to "hero"'s repairing audio and configure the best sample rate for recording.
My preferences Audio Hardware on Audition are:
Device Class: CoreAudio
Default Input: Onyx Firewire (0025)
Default Output: Onyx Firewire (0025)
Clock Source: Device
I/O Buffer size 512 Samples
Sample Rate: 96000 Hz
Question 1: Expert said do not use more than 48000 Hz for recording voice only and use mono. Humans can't listen more than 20000 Hz... well... My recording on 96000 Hz is better than 48000 Hz and I'm not Superman... and can hear better difference with 96000 Hz recording. Don't know why.
Question 2: Why after selecting Audio Hardware preferences to 96000 Hz after recording the File Panel said my track is 48000 Hz Mono 32 (Float)? My hardware can record at 96000 Hz and preferences is set correcty at that Bit Depth then WHY don't appears 96000Hz on File Panel after recording and instead appears 48000Hz?
Question 3: Don't understand the differences of my 24-bit/96kHz Onyx FireWire I/O and 32 bit (Float) of Audition. If I record using 24 Bits (Maximum bit depth of my hardware) why appears 32 bit (Float)? My recording are 24 Bits or 32 Bits?
Thanks :-)
Tom

CS6 has a serious issue with saving files correctly. The program is asuming that 48kHz is the maximum you will be using and in my case it saved a 96kHz recording with a 48kHz internal header. The file size is consistent with all my previous 24/96 recordings and it sounds just fine interpreted correctly - but played an octave low in frequency and tempo it really sucks unless you are a Blue Whale..
I can play it "interpreted" as a 192kHz file just fine, and it now sounds 100% right, but I cannot save it correctly. I have yet to find out how to recover because when I use "convert Sample Type" it saves it with the same mistake - the wrong sample rate off by the same ratio again.
It is a program flaw - so at this point you cannot record with sample rates higher than 48kHz in CS6 and depend on your file being OK.
Tom the reason you can tell the difference between sample rates is that your hearing has two dimensions, frequency and timing, I sure hope you can't hear the bats at night but I expect that you can tell a good drummer from a bad drummer. In addition there is also the issue (dimension) of bit-depth - instrument decay and acoustic space occupies the time space between notes and if it is not sampled at the right time and place or rounded off to the nearest digital digit you have a problem.
You all know that some humans have perfect pitch and others dont, this gives you some indication how much we each differ. Some people have even learned to use echolocation; the best know cases being blind people because they are not supposed to be able to find their way and know where they are. You can learn underlying concepts of this little discussed aspect of human hearing here and hit the university libraries for the rest. http://en.wikipedia.org/wiki/Human_echolocationhttp://
SteveG's comment are only true in one of the three dimensions - frequency range. We all hear about the same for starters anyway.
The second dimension, timing, is what provides spatial information. Coarse sampling effects this as well. My most recent experience was when transcribing some old casette tapes - when I experimented with FLAC and WMA lossless I found that they din NOT downsample well. Spatial information was significantly deminished be that aucoustic or studio work. Needless to say this surprised me because I was auming that I could drop the samplerate to 48kHz as soon I was done editing and save a lot of drive space. For now I made some excellent 48kHz/24 bit mp3s (320) because they altered the sound the least.
Now you know some additional reasons why older guys like me who have lost the high end and a lot of decibel as well can still tell the difference:we have good sense of timing.
Anyway - I need to learn how to edit my files to reset the sample frequency header - real fast. I just recorder a fabulous Madrigal group for 2 hours and my files are lost in the vortex.

Audition sample rate mismatch

I have been using Audobe Audition 2.0 for a number of years, and now have upgraded hardware and software (Windows 7 running on a new good spec 64bit Dell), with the intention of running CS6.
Unfortunately when I go to record, using the same linein as previously, I get the following message:
"Sample rate of the audio input and output devices do not match. Audio cannot be recorded until this is corrected."
I think I know what it means, and have ensured (as far as I am aware) that the sound card is set to 44100, 16bit, the same as the settings on Audition, or appear to. So what have I missed. (I do know XP better than 7).
Many thanks
Peter G

You also have to set the sample rate in the Windows Sound control panel. Select your audio source in the Windows Sound Recording panel and click on Properties. GO to the Advanced tab to make sure that the sample rate is set correctly there as well as everywhere else. Hopefully that should sort the problem. This is an extra vaguery of Windos 7 audio drivers.

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.
Is there anything I can 'easily' do to decrease the sampling rate down to a low level? Everything I have tried doesn't seem to work and I don't really want to go away from SignalExpress.
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

Setting Sample Rates in Audition

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