Sample Rate 44.1 kHz creates Hum/Buzz in speakers

I noticed a quite hum in my speakers, Mackie MR8s, and cold not figure out what it was. So going through the menus in Logic Express 8 I found the "audio" under settings. Where it says sample rate I changed it to 48 kHz and the hum disappeared. I went to switch it back and I get a buzz, but only when the 44 kHz sample rate is chosen. Does any know why it would create a hum? Thanks.

It's more of a a real quite hiss than a hum. My battery backup was making them buzz so I've moved that across the room. Could it because of their close proximity to my imac or other electrical items on the other side of my wall in the other room. The hum doesn't come through on headphones or in recordings. Thanks.

Similar Messages

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.

Could you confirm that ALAC doesn't play sample rate of 96 kHz?

Hi,
Could you confirm for me that ALAC doesn't play sample rate of 96 kHz? Because my codecs converter, regardless of what sampling I set up gives me 44.1 KHz in ALAC.
Thanks

Newme wrote:
Ed 2345 wrote:
"Unfortunately, if you are working with hi-res audio files, Apple Lossless is not lossless."
Really? Can you back up that. It should be easy to prove. if it really is so. What kind of files don't fulfil that?
Newme,
Simple enough to demonstrate. If you take a WAV 96/24 and use iTunes to convert it to Apple Lossless and then back to WAV (using "Automatic" setting in both cases), you end up with a significantly smaller file. Something gets lost.
Asking this because DSD files can't actually be tuned into other formats without losing data. Whether there is actually anyone that can hear the difference is another matter of course :-)
Wasn't referring to DSD, but it is an interesting question. The usual lossless formats, ALAC and FLAC, are designed to improve on the inefficiencies of PCM encoding as is used in WAV, AIFF, and audio CDs. DSD is a horse of a different color, as it is not PCM based at all.

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

Can anybody recommend me on the strain gauges which are attached along a steel bar, say, 3 mm in diameter, and can respond at a sampling rate of 100 kHz?

Or anybody has similar application, it would be very grateful if you can share your experience.

Many thanks for your information. We are about to investigate the dynamic performance of soil anchorage subject to dynamic impulse in a geotechnical centrifuge. Basically, 5 model anchorages (3 mm in diameter) are to be constructed. Each model anchorage is instrumented with 1 accelerometer, 1 load cell and 6 strain gauges. Therefore, the minimum instrumentation would be
1) 30 strain gauges
2) 5 accelerometers
3) 5 load cells
It is worth noting that all the instruments would eventually reside in a geotechnical centrifuge operating in a 10g gravitational field. Whilst the DAQ system would experience much less 'g force' since they are to be placed close to the rotation axis of the machine.
These model anchorages would be tested one by one dyn
amically on each individual basis, whilst the other four are also preferably to be monitored during the tests. For an active model anchorage which is under testing, simultaneous sampling is preferred for its 8 sensors to be measured at the same instant in time (Besides, it is also understood that the natural frequency of the model anchorage is estimated to be in the range of several kHz). For the other four model anchorages, simultaneous sampling could be less demanding. It would therefore be ideal that simultaneous sampling can be switched among the active testing model anchorages.
I am greatly appreciated if you can further advise me on this. Thanks in advance.

Sample rate error

Hi everyone!
This is my first massage in this great forum. I'm a beginner in LV so be patient, moreover I apologize for my poor English... Anyway I hope to be clear.
I have to read different signals from different sensors such as pressure sensors and speed sensors. Both of them give to my SCXI module (SCXI-1303) voltage signals. I aquire them with PCI-6220 M series board.
At the beginning I created a task for each signal (using MAX)... They worked properly at the sample rate of 1 KHz only if not running at the same time. I later discovered that more tasks of the same type can't run simultaneously on my board. So I have later created a global channel for each signal and then I have created a new task adding the channel of analog input (voltage type) in it (always using MAX). I chose as sample rate 1KHz as before but now there's an error:
Error -200332 occurred at DAQ Assistant
Possible
Reason(s):
Specified sample rate is higher than the fastest rate
supported by the device.
Property:
SampClk.Rate
Corresponding Value: 1.0 e3
Maximum Value:
480.769231
Why? Where is my mistake?Please help me.
Thank you!!
p.s. I use Labview 8
I love the smell of napalm in the morning

OK.
This information certainly helps.
As you can see in appendix A page 1 of the 1100 manual the settling time of the 10KHz filter is around 200 us. A little math leads us to 500Hz which is really close to your value. The problem is that the 1100 module has only one filter shared between channels.
This filter needs a bit of time to settle to the correct value while switching from one channel to the other, clearly if you don’t scan between channel you don’t occur in this problem.
I don’t know anything about your application, maybe you can circumvent the problem running one task at a time and closing and opening each task with a ms delay (Typical software time)
Kind Regards
FiloP
It doesn't matter how beautiful your theory is, it doesn't matter how smart you are. If it doesn't agree with experiment, it's wrong.
Richard P. Feynman

Change the audio sample rate?

It looks like upon export, the audio default sample rate is 44khz, which creates some issues when uploading to vimeo and youtube. Is there a way to switch the sample rate to 48khz?
Thanks!
Kyle

Are you exporting from Motion straight to Vimeo or Youtube? If so, Motion encodes the audio at 48khz.
I've never had any problem using 44 or 48 kHz when sending files to youTube or Vimeo. Please explain the kind of problem your having... and your workflow.
ps... You can send to Compressor from Motion and change the sampling rate. Along with that you can send to youTube or Vimeo from Compressor.

Sampling Rate in A

I'm unable to set the Sampling rates to any value other then 48 kHz using the ASIO APIs. The default setting is itself 48 kHz. I could determine this using ASIOGetSampleRate(). When I try to set it using ASIOSetSampleRate(), it returns ASIOERROR.
How can I get around this problem?

shyam wrote:
Thanks jutapa. I was indeed working with a creative card(SB A2Zs).
Is there any way we can get direct support for sampling rates below 44. kHz using the X-Fi's?
-Shyam
I suppose it's not possible for ASIO drivers with Creative cards (if the modes are hard locked like the 6/48 and 24/96).
Most Creative cards supports sampling rates of 8, .025, 6, 22.05, 24, 32, 44., 48, 96 (and 92) kHz ... supposeingly, you can use lower rates only with DS/MME/WDM drivers.
If your intend is to get windows kmixer bypassed then you can also use WDM/KS instead of ASIO. I think you can work with /KS just the way WDM allows. You find more specific info from MS Support.
There are also cards like Lynx22, which allows you freely set the SR from 8 kHz to 200 kHz, including all standard rates with variable adjustment. This card is ofcourse a bit more expesi've vs. Creative cards. Lynx22 street price is around ~650-700$.
jutapaMessage Edited by jutapa on 0-25-2006 09:03 PM

Changing audio recording depth and sample rate in QuickTime Pro?

Last week, I set up QuickTime Pro 7.4.5 to record audio in 24-bit, 48 KHz format -- only now, I can't remember how I did that. I think that I may have used some other software to change the settings of my Mac Pro's built-in sound card. Does anyone know how to do this?
I made several such recordings, just last week, as a test, simply using a microphone plugged directly into my Mac Pro, and using QuickTime Pro as the recording application. During playback of such QuickTime audio files, I can hit <command-i> to see that the recordings are indeed 24-bit, 48 KHz. But now, when I repeat the same process, my QuickTime audio recordings are made with 16-bit resolution, at 44.1 KHz. I remember putting the settings back to those values after my recording session of last week. But I can't remember how I adjusted those settings. Does anyone know how to do that?
Thanks.

Thanks for your response.
I will use Apple MIDI Setup, as you suggest.
I want to change the sample rate to 48 KHz, as the final destination is DVD and I don't want to convert from 44.1 to 48 and thus suffer a possible loss in quality.
As for bit depth, I want to sample at 24-bit, since I'll be processing the sound afterwards and don't want audible rounding errors during the number "crunching". Afterwards, I'll just drop the eight least significant bits, or perhaps round them up or down (to zero or 256).

Audio Sample Rate

Can I change the audio sample rate of a clip (imported from camcorder) "after" it's been imported into Final Cut X?

Thank you for the quick response. My video and audio are out of sync (imported from mini-DV camcorder). The audio sample rate is 32 kHZ, and I believe I should have used a "custom" sample rate setting of 48. I really don't want to start over with this project and re-import all of my footage, but that may be the best thing to do. I also learned from a previous post that I should capture no more than 10 mins. of footage at one time.
P.S. I'm about done with this camera!

Audigy 2 platinum sample rate problem

People i need help with my audigy... Its connect to my Ht sony ddw880 via coax cable... In audio console, when selected spdif digital out (PCM) sample rate for 48 khz... everything goes fine..But when i select sample rate for 96khz, my receiver detects the signal as 96 pcm but theres no LFE signal... The sound plays normal but without subwoofer... Ive contacted sony and the tell me thats a problem with the signal that the pc is outputting in 96khz mode... cause in 48 khz its ok... I need help with this problem... please anyone.....!!!Thanks a lot.. PS: Sorry about my english (Brazilian)

People i need help with my audigy... Its connect to my Ht sony ddw880 via coax cable... In audio console, when selected spdif digital out (PCM) sample rate for 48 khz... everything goes fine..But when i select sample rate for 96khz, my receiver detects the signal as 96 pcm but theres no LFE signal... The sound plays normal but without subwoofer... Ive contacted sony and the tell me thats a problem with the signal that the pc is outputting in 96khz mode... cause in 48 khz its ok... I need help with this problem... please anyone.....!!!Thanks a lot.. PS: Sorry about my english (Brazilian)

Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000. Is this possible in Audition ? I have been going through all the preferences and all the tabs and I can't find this option. All I find is a convert option, or the adjust option. But that's not what I want. I want to mix down this way.
The closest thing I found is when I go to "Export Audio Mix Down", I can select 32 bit. Then there is a box for sample rate, with all the different values. But it won't allow me to change it from 44100.

JimMcMahon85 wrote:
Can someone explain this process in laymens terms:
http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
Second point: you cannot dither a 32-bit Floating Point recording, under any circumstances at all. You can only dither a recording if it's stored in an integer-based format - like the 16-bit files that go on a CD. Technically then, you can dither a 24-bit recording - although there wouldn't be any point, because the dither would be at a level which was impossible to reproduce on real-world electronics - which would promptly swamp the entire effect you were listening for with its own noise anyway. Bottom line - the only signals you need to dither are the 16-bit ones on the file you use for creating CD copies. And you should only dither once - hence the seemingly strange instructions in the Ozone guide about turning off the Audition dither when you save the converted copy. The basic idea is that you apply the dither to the 32-bit file during the truncation process - and that's dithered the file (albeit 'virtually') just the once. Now if you do the final file conversion in Audition, you need to make sure the dithering is turned off during the process, otherwise all the good work that Ozone did is undone. What you need to do is to transfer the Ozone dithering at the 16-bit level directly to a 16-bit file proper, without anything else interfering with it at all. So what you do with Ozone is to do the dithering to your master file, and save that as something else - don't leave the master file like that at all. After saving it, undo the changes to the original, in fact - otherwise it's effectively not a file you could use to generate a master with a greater-than-16-bit depth from any more, because it will all have been truncated. Small point, but easy to overlook!
just what's the easiest way to test if a simple dithering setting is
working for 32-bit down to 16-bit in Audition?... Why is there no info
about dithering from 32 bit to 16 bit (which is better then dithering
from 24-bit isn't it)?
I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
It's all about educating people in the end - and as you are in the process of discovering, all education causes brain damage - otherwise it hasn't worked.

How to create a wav file from 24bit 96Khz sampling rate data

Hi
I am trying to make an VI which will play sound while acquiring data from PXI 4472 DAQ card.
My sampling rate is 96Khz and PXI 4472 card is a 24bit card.
Wave files are in 8 or 16 bit and the sampling rate is 8000, 11025, 22050 and 44100. How will I be able to play the data which I am acquiring.
How would i normalize the data into the required format needed for most of the sound cards to play.
Or are there any codec available in Windows XP which i call to play a 96KHz 24 bit sample
Does anybody ever encountered this type of problem.
Thanks in advance
Nitin

Whilst the 'standard' RIFF format specification usually accomodates 16 bit data, there is of course no reason that you can not create your own extension. It just won't be playable by Media player using the 'standard' installed drivers or codec. This may not be a problem....
WAV files can and do support other formats, you just need to know how to handle them......
There is howerver a 4GB limit (related to the pointer size in the WAV specification) which with higher bit depths on the sampling does start to become a bit of a problem.
To give you a few samples of other types of wav files check out the following site here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
The following definitions for WAV audio formats may also be of interest here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
Good luck with 24 bits.

X-Fi Elite Pro: Can't Insert FX with 96 kHz Sampling Rate

I can't seem to Insert FX with 96 kHz Master Sampling Rate set.
"Unable to load effect. .... user lower Master Sampling Rate or unload effects"
but there are no extra effects loaded (except default Reverb and Chorus aux effects).
PIV 2.8 GHz? 52 M RAM

Hi Daniel,
Thanks for taking the time to reply to my post. Here are my hardware specs:
CPU: AMD Athlon 64 X2 3800+ 2.0G
Motherboard: Foxconn <FONT face=Arial>NF4SK8AA-8KRS NF4SLI
Memory: WINTEC AMPO 52MB 84-Pin DDR SDRAM DDR 400 (PC 3200) Unbuffered (2GB total)
Power Supply: COOLMAX CW-650T (650 watt, SLI capable)
Floppy Dri've: MITSUMI .44MB
Video Card: (2) GE Force 6800XT running in SLI
Before I attempted to install the X-FI Elite Pro, I did have a problem getting the video card to work, but I found that the solution to that was to change the voltage setting on the power supply. After I solved this problem, I tried to install the soundcard, which as you can see has opened up a new issue.
The beeps are constant, and the computer looks like it is attempting to reboot with each beep.
I'm not sure what the power supply is rated for. What do you mean by that? I know it's SLI capable whatever that means.
I'm not sure if the BIOS has power settings for the PCI bus. If it does, do you recommend a particular setting?
Thanks for your help.

How do I capture at an audio sample rate of 44.1 KHz?

Hey,
I'm wanting to capture some footage that's been recorded at a 44.1 KHz sample rate. Presets on FCP 5 only have 48 KHz.
How do I change this?

I just checked the import settings on two other editing systems I use... Canopus DV Storm II and the Matrox RTX 100 Extreme. Only the Canopus has a preset for 44.1k import. Even Adobe Premiere, using the native IEEE 1394 connection, doesn't have it. I am not sure if FCP has a "CUSTOM" import dialog that might allow you to manualy select?
If you have found this information helpful, please select the icon for this. If it helps you resolve this issue, please select "answered".

Sample Rate 44.1 kHz creates Hum/Buzz in speakers

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