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

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Audible glitching - audio sample rate mismatch?

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I'm using a sequence with the DV-NTSC preset with 48KHz audio. I made sure to set it up in Easy Setup beforehand so it would be the same as the sequence. The camcorder I'm using for capture is my old Sony TRV-70 (have also tried my TRV-50). I can't get either camera to tell me what sample rate the tape was shot in; all i can get it to tell me is 16 bit and SP mode. I'm assuming this will be 48KHz?
When i try capturing with the DV 32KHz easy setup (and sequence) it gives me the error about the sequence rate not matching what's on the tape, and it sounds worse.
I don't know what camera originally shot the tapes. But the audio sounds perfectly normal coming out of the speaker in my sony camcorder.
Am I missing something?

16 bit and SP mode. I'm assuming this will be 48KHz?
That's correct.
Make sure the Sequence you select is a 48 kHz choice abd double check that you do not have a PAL/NTSC mix up.
Al

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I would like to know how to take manual control over core audio regarding sample rate destruct... er conversion. First - I know the workarounds - simply closing the audio player of choice, resetting the external hardware and relaunching the audio player of choice.
setup:
I run external converters with an external sample rate clock source. No problems or issues here. I keep my music collection segregated by SR (96k, 88.2k, 48k, 44.1k) as I ALWAYS listen through external converters.
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What I want is an indicator that will tell when SRC is turned on and further, what the input and output sample rates are (as far as core audio is able to determine from the hardware anyway). In my world this would have been a check box in a preference setting. Perhaps someone has written a script or app for this? Command line instruction?
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Start with http://developer.apple.com/documentation/MusicAudio/Conceptual/CoreAudioOverview /Introduction/Introduction.html and direct further queries to the developer forums under OS X Technologies.

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In my application i need to play the audio with jmf player with sampling rate 48000hz. but i found that jmf player plays the audio with sampling rate of 44100hz only.but my application needs to play the audio with sampling rate of 48000hz .please help me how to do this using jmf .
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hi,
In my application i need to play the audio with jmf player with sampling rate 48000hz. but i found that jmf player plays the audio with sampling rate of 44100hz only.but my application needs to play the audio with sampling rate of 48000hz .please help me how to do this using jmf .
thanks in advance,

Problems with audio Parallels sample rate

Hi!
I recently I installed Finale 2014 in my iMac (Os X 10.9.2) and, when I try to run the application, I receive an error message telling that I must adjust my audio sample rate to 44.1 kHz and Finale crashes. So, by going to Applications --> Utillities --> Audio and MIDI Setup, I adjusted the sample rate to this value and the issue continues to appear. By contacting Finale support, they told me that I must remove my audio device from Parallels Desktop 9 (I was not allowed to adjust this device). But, the problem is that I simply can not remove the device - when I select the device, the " - " signal is in gray. How can I remove the device, in this case?
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Hello!, first of all thanks for your reply. Yes, the camera is the same, the problem is on my computer, I have discovered a solution: first I capture the audio, only the audio, and then the video, only the video. After this I soncronize the two separate files on Final Cut timeline. It´s a slow process, but I am really desesperated. I don´t know what happens, I few minutes ago I have captured a 30 mins. video and the sample rate of the audio was 42.145k (???). It´s a crazy thing...

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:
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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?

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Hi,
I have USB capture device which supports HDMI input and I wish to use it to connect to my digicam and stream it out with FMLE. While I can see that the video input is working properly, the audio is not working. The error message "Problem with audio capture device - Please verify that the audio device is working correctly and is not already in use." always appear when I try to select the device's audio device. Using the graphedit, I can see that the capture device's audio output pin only supports stereo 16-bit 48 kHz PCM output. As such, can I confirm that that FMLE only support 44.1kHz (as well as other 44.1kHz divisible) sampling rate?
Thanks
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After researching and playing with the system, I found a solution. Uninstalling the "Realtek High Definition Audio Driver" under the Control Panel -> Hardware and Sound enables sampling rate up to 384KHz instead of 48KHz.
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I have a huge problem with my 2009 iMac's digital input that I didn't run into on my older MacBook running the same OS (10.6.x), and it's driving me nuts!
Basically, I'm trying to record a digital feed from an ADC hooked up between my hand-built tube amps/turntable and my iMac, with the goal being the transfer of some of my LP records to 48/24 LPCM for archiving and editing. The ADC's output rate is fixed at 48000Hz and cannot be changed. My iMac, however, refuses to lock on to the device rate, instead forcing 44.1kHz in the audio/midi setup. In fact, I can't get it to do anything but 44.1kHz on anything I plug in (even my DAC's passthrough) despite the fact that I need to use 48kHz. The audio is correct for a few seconds, then forces back to the wrong rate again. No matter how many times I try, the computer insists on using the wrong sampling rate, which has rendered impossible any hope of using my iMac as an audio recorder.
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Seeing the exact same thing here. External digital source is 48kHz, and this Mac Pro Desktop will only stay on that setting very temporarily (for about .10 or .15 seconds) then it reverts back to 44.1kHz. Even if I don't have a signal or even a Toslink cable plugged in to the input.
(I'm surprised you are getting 2 or 3 seconds of correct audio, when I'm getting about 1/10th that before it reverts back to the wrong sample rate. Maybe related to a buffer size option somewhere?)
I have tried every known option to reset or re-configure this, and even to the point of creating a new user, resetting PRAM, resetting SMC, etc. I was going to try upgrading to 10.6, thinking maybe it was a 10.5.x problem, but now I'm not sure that will help.
Had I the choice to set the setting to be 48kHz and lock it in, it should stay on that setting - (even if the chance exists that what I'm getting is not correct.)
More often than not, I know what I'm doing more than Mac OS does, and I want to force it to do what it needs to.
There really needs to be a "do it because I told you so" option for most of the Mac audio configuration system. This automatic "the hardware decided to do something for you" stuff is absolutely horrid for a system that is marketed as "Pro". A real "Pro" system would let me edit the configuration once, with the tool (text editor hopefully) of my choice, rather than attempting (and failing) to automate things.
+1 for another instance of this being "a huge problem" and "it's driving me nuts!"

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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

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Hello, I am new to Logic/DAW recording so please forgive this "newbe" question.
I was getting an error "while trying to synchronize audio and midi" after installing an audio loop.
I found that the "fix" was to go into the settings>audio>samplerate>48K
Fine, the error message stopped.
But, does this mean that if I wish to use the audio samples in Logic8, I must use the 48K sample rate for the whole project? Or can I use a higher sample rate for live audio (ie 96K) and use the L8 samples at the same time?
Thanks for any help you may offer, Brian.

Hi, and Welcome to the forum
instead of changing the sample rate of your song to match your loops, change the sample rate of your loops to match the song. Read pages 143-145 of the manual
Don

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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.

DIgital audio in sample rates, 441 48 96 but no 882... is there a way to ad

When feeding my macbookpro a digital audio signal via optical cable while viewing the audio midi setup utility, it shows that the mbp's internal clock automatically switches to the incoming sample rate. I got it to work with 441,48 and 96k. When I fead it the 882 sample rate it automatically switches back to the mbp's internal clock and reverts to line input instead of digital. So the audio midi setup utility shows when it is in digital input mode the three sample rates 441,48 and 96k in the format pop up window and not 882 so it must be just a software issue and I am wondering if there is a way to add the sample rate to the list. If it can take in 96k it should also be ale to take in 882. So does anyone know how you could ad a sample rate of 882 to the list in the audio midi utility format list?

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I received some audio files for a project that were accidentally sent at 48k instead of the 44.1 I need.
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Import Audio File (Shift Command I).

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Thanks for the response. My project currently does have those options checked, and it's still not converting.
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Blue lines through audio, NOT sampling rate

Hello,
A friend is editing some audio, using a lot of tracks (around twenty) and shifting the speed. For some reason audio clips that were slowed to 80 percent are appearing with a blue line through them and playing at a very low volume. I checked the sampling rate and it's 48kHz. Also frustrating is that these playback issues are intermittent.
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thanks,
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He probably doesn't have the CPU horsepower to render twenty audio tracks on the fly. In User Preferences, you can do a few things to reduce the need to render audio:
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b. reduce the audio quality to low - it really doesn't sound bad at all, and it only affects your preview audio;

Audio problem ( sample rate/ mismatch samplerate )

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