Where is select sample rate?

I can't believe that I don't find the sample rate config in logic 8. Where in the menu can I chose sample rate? I have a RME firecafe 800 and would like to record higher than 44.1.
/Erik
<subject edited by host>

You can also choose to have it appear in the transport at the bottom by right-clicking on the transport and selecting the customize option.

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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.

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)

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

The selected combination of bit rate and sample rate is not allowed.

I've always burned my purchased CDs into iTunes using a AAC 320 kbps and 48.000kHz. Now when I select that option in iTunes 9 I'm getting a "The selected combination of bit rate and sample rate is not allowed." pop up screen. So is it impossible to burn my albums in a 320kbps/48.000 AAC format?. I had no problems in previous versions of iTunes using this setting. I guess I might have to switch over using a MP3 import setting instead.

Audio CDs are encoded with a sample rate of 44.1 kHz. Ripping them at 48 kHz requires resampling and does not improve the quality. Unless you have a special need for 48, use 44.1.

Can't find where to change audio sample rate

I'm trying to change the audio sample rate but can't find where to change it

You can change it in SoundTrack Pro, or with QuickTime Pro.
-DH

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.

Selecting sample clock frequency for NI 4472

HelIo:
I have been using NI 4472 DSA on a PCI slot for reading accelerations. I was reading the NI 4472 User Manual and under "Selecting your sample clock frequency", it specify two increments (95.37 uS/s for Fs<51.2 kS/s and 190.7 uS/s for Fs>51.2 kS/s). Can you explain what these increments mean? I understand the sampling frequency, but what does this time increment mean?
Also, it says "If you do not specify a rate at a multiple of the increment, NI-DAQ will automatically choose the next higher step for you". Dose that mean I cannot specify any random value for Fs? Dose it have to be a certain multiplication. Please explain this for a novice to Labview/DAQ.
Thanks a lot in advance.

Hello MiamiVipul,
Those statements in the User Manual mean that the PCI-4472 only has a finite set of sample rates that it can produce. The allowable sample rates are given by 51.2 kS/s - k * 95.37 uS/s or 51.2 kS/s + k * 190.7 uS/s, where k is an integer. As you can see, the step size is quite small, so you still have a lot of flexibility.
As an example, if you specify a sample rate of exactly 51.2 kS/s, that would be fine. However, if you try to set the sample rate to something between 51,200 S/s and 51,199.990463 S/s (51.2 kS/s - 95.37 uS/s), the driver will automatically coerce that sample rate to next highest allowable rate, or 51.2 kS/s. In order to find out the exact value that the driver is using, you can read the from the DAQmx Timing Property "Sample Clock:Rate" as shown below:
I hope this helps!
Best regards,
Message Edited by Jarrod B. on 01-04-2007 09:03 AM
Attachments:
ReadSampleRate.JPG ‏6 KB

Changing the Sampling rate

Hi,
May I know how do I change the sampling rate on a Thermocouple node dynamically through a VI on the host ?
I notice there is a Config node in the "Start" case where I can set the "Sample Interval Mode" to "Host Driven" but I do not know how to continue after that...
Thank you very much !
Regards,
Xinke

A user message is created by simply right-clicking on the node and selecting "Create User-Defined Variable." Once you set it up, that variable will update over the network automatically.
Once you've got that setup, you need to figure out how you are going to use that variable. You should probably start here: The LabVIEW Wireless Sensor Network Module - Under the Hood
Note: User-defined variables are only available in LabVIEW 2011 for WSN.
Chris
Certified LabVIEW Architect
Certified TestStand Architect

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.

Convert Sample Rate on Import- does not work

Setting: *"Automatically convert sample rate of imported audio"*
You'll want to set this if you have audio on your computer
that wasn't sampled at 44k. It prevents your imported audio from
sounding like Mickey Mouse drinking a gallon of espresso and inhaling helium.
HOWEVER...It does NOT work when you drag audio into the project from
Logic's internal File Browser.
Dragging audio into the project from Logic's Browser,
or browsing to the file in Logic's Browser and choosing to
"Add selected Files to Audio Bin",
will NOT convert to the audio's sample rate.
Sample rate of your audio is only converted if you do: "Import audio file",
then browse to the file(again) in a popup browser dialog, and then choose import.
If the audio is already in your Audio Bin, choosing "Convert File"
will bring up another Finder Dialog asking you where you want the
converted audio to be placed.
Now you'll have to Browse (yet again) to the song's folder, because that's where
you'll probably want it anyway, and choose to put
your converted audio into the song's folder.
Please realize: If a composer or producer wants his/her audio to sound like
Mickey Mouse on Helium and Espresso, he will do it after the fact, on purpose, using a plugin.
Is it too much to ask, to have this Setting do what it says it does, with ALL audio brought into the song?
No matter how I bring audio material into a Project, the sample rate should always conform
to the Project's sample rate.

Hi there
I've been using Captivate since version 1. And to be honest, I'm not sure I've ever seen the option work. If it does work, I'd also like to know under what conditions it works. Perhaps it only works with certain applications. I'm not sure. Largely I've ignored the option exists because I've never recorded anything where I wanted tooltips to be an option. I've never wanted to simulate to that detail.
You should probably consider filing a bug report.
Cheers... Rick
Helpful and Handy Links
Captivate Wish Form/Bug Reporting Form
Adobe Certified Captivate Training
SorcerStone Blog
Captivate eBooks

Audio problem ( sample rate/ mismatch samplerate )

Hi everyone,
I'm using a Macbook white Mid 2010, everything is fine but recenly i bought a USB interface to record music from my instruments ( Mic and guitar ) through garageband..
I have watched steps on how to connect guitar to garageband, and i did those steps by selecting USB channel as input..
The problem is every time i test my guitar or microphone, it sounds terrible just like little chipmunk or something. Some ppl told me my sample rate is mismatch. I also was checking around on Audio Midi Setup, i can't change my sample rate for my USB audio interface. The input for it, is 28000KHz, and i was told garageband only accepts 44000KHz.. So i thought my USB interface was having a problem..
However today i have tested my microphone on my old iMac white, i did plug in my USB interface and microphone, it worked so fine. The sound was not like on my macbook. it just fine and very clear. So my USB interface is not the problem..
BTW audacity also works fine on my macbook, only garageband..
Could someone help me? I can give you more details about it..

Canon cameras are notorious for recording with nonstandard sampling. Usually, though, material will capture without synch problems.
Make sure you have "abort capture on dropped frames" and "create new clip on timecode break" enabled in fcp: user preferences: general and do not capture over a control track break (where you see just "noise" in the video

Core Audio - Sample rate

I've just got myself a copy of Logic Pro 8 as a complete newbie and have hit a bit of a hurdle within 24 hours of opening the software.
I was using the software without a problem just tinkering around using a Carillion MIDI interface to play some software instruments. I don't have a mic yet so thought I'd try my bluetooth headset just so I could play with recorded vocals. When I started speaking while Logoc was recording it came up with an error message
"Core Audio
Sample Rate 8000 not allowed."
I then turned off my bluetooth headset and turned bluetooth off on my Mac so I could go back to using the MIDI keyboard but got no sound. I closed Logic and re-opened, selected "Empty Project" withing the "Explore" tab and was again presented with a series of the messages, all reading as per the above error messages but I get 3 with "8000" and one with "73536".
When I'm playing the keyboard I can see the note I'm playing or the chord in the dialog box at the bottom of the Logic interface screen but again, no sound.
Any help would be hugely appreciated. Again, I am a COMPLETE Logic novice so please be gentle with me.
Regards,
Mark

Hi Jounik,
I went into Audio MIDI Setup and the headset didn’t appear. The keyboard appears as “Midilink” but I couldn’t see where to check the sample rate. I did however use the Test Setup facility and the keyboard was only producing a sound when the key was released.
I also had a look at: Logic Pro > Preferences > Audio (Devices) but wasn’t really sure what to look for. This is what I was confronted with http://adoseof.co.uk/resources/Picture1.png
The project sample rate was set to 44.100 KHz
I like the idea of using the internal mic. I’d tried that before but was getting feedback. Plugging the headphones in seems like a good solution.
To try and get the keyboard working again I opened a new Empty Project and left the dialog with the default settings as indicated here: http://adoseof.co.uk/resources/Picture2.png
I’m not sure what you mean by “In the I/O slot of the channel strip open a synth, e.g. ES1”
I then used the on-screen keyboard and still no sound was coming out.
Any other ideas?
I tried the keyboard in garage band and it worked fine so I can only assume it must be something to do with Logic.
Mark
Message was edited by: hotsawz

Converting a song from one sample rate to another

i recently had to convert a song that had been done in 44.1 to 48. that is, all the audio i had recorded or imported and chopped up was at 44.1. what i wanted was to change them all to 48, so that the new recordings that had to be done would be at the new sample rate. obviously i can select copy and convert in the audio editor but it doesn't change the region file referneces in the arrange.
short of actually physically making sure that logic can't the old audio files and thus forcing it to change file reference, i had to change each audio file with sample rate converter maually.
surely there must be a better way. anyone know what it is?

o.k. yes, It's all coming back to me now...
The work around is a little odd, but the way I did this before, was after batch converting the audio files to the new sample rate, I had to literally throw the original audio files into the trash, and then empty the trash. ( Yes, I backed up first, just in case... )
I then opened the Logic song, and a prompt says "X(amount) invalid regions changed." Then Logic begins to create overviews for all the new audio files, and then it's done. The song is playing all the new sample rate converted audio files, where the old ones used to be.
You need to empty the trash though, or Logic goes and finds the original audio file in the trash, and uses them.

External sound card sample rate issue

After installing Yosemite I am having problem with Sample Rate settings on my external audio interface. It was working perfectly before the upgrade.
The issue is that the sample rate on my system is set to 44.1 Khz. When a new application attempts to access the audio card it plays but breaks up. This is indicative of a sample rate issue so I tried to set the rate to 48 Khz and then back to 44.1 Khz. Once I do this the system works properly until another application tried to use the card and I have to repeat the same process over again. While it DOES work it is not correct functionality and there is obviously something wrong.
At first I thought is was just my system until my friend also commented on it happening to him too.
I am using an RME Fireface UFX with the latest drivers issued on October 8th of this year.
My friend is using an M-Audio unit and he is finding the same issue.
I have already checked everything in Audio / Midi Set-up control and reinstalled the drivers for my audio interface. I also looked for any new settings to Yosemite to see if there was something new that might affect this. No luck and no leads on where to look for the next option.
Anyone have any ideas?

Hi,
The sound card do not support all sample rates. Which sample rates your sound card support you will find in the the manual.goto help in the toolbar select find examples and search for sound Open the Sound Input to File.vi file.This will give you a template for recording sound.You have to set the sample rate then open a sound file for write.
Please post a sample VI that you have created.
also,please go through this:-
http://www.zeitnitz.de/Christian/waveio
Thanks as kudos only

Where is select sample rate?

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