Changing sample rates for recorded audio?

I've been trying to record VO through premiere. I'm using an m-audio mobile pre which has never given me trouble in any other application. When i turn on Meter Input(s) Only from the panel menu of the track mixer i can clearly see the input move in time and responsively. When I record however the resulting clip is much shorter than the ammount of time recorded and the audio is all garbled and chopped. It clearly jumps out to me as a sample rate problem. I looked at the clips and they are 44.1 but the sequence i'm recording to is 48. I made a 44.1 sequence and recorded to that which worked fine but there must be a way to change the recording sample rate right? I can't find a solution anywhere. Anybody know?

i'm just following up here. Thanks for the replies. I understood i would lose data as one answer suggests, but i wanted to convert it so i could work in a new song environment in which all the NEW tracks were recorded at 192. So the second answer was very helpful. incidentally, with further research we are recording at 96.

Similar Messages

Can't change sample rate for digital input on Mac Pro

Hello all,
on my Mac Pro with 10.4.10 I can't change the sample rate for the digital input. Whenever I choose 48000 or 96000 Hz it returns to 44100 Hz after a few seconds. Feeding a 24 bit/96000 Hz signal from an external ADC into the optical input doesn't help. No input signal is available for digital recording software such as Sound Studio or Cubase. After changing the sample rate in audio midi configuration to 96000 Hz the sound can be heard for a few seconds but when the setup returns to 44100 automatically the signal is, of course, lost.
I've deleted all relevant preferences and restarted with resetting paramter RAM. Still the same. With external hardware such as M-Audio Firewire equipment setting the sample rates works properly.
Thanks to all for helpful clues.

Hi,
when E&M signaling is configured on digital interface like the VWIC is, 2 or 4 wires operation is not applicable because there are no wires at all, and reported only for compatibility with the analog E&M card.
Consequently, you cannot configure that and it will not make any difference to effect of the connection.
Please rate post if it helps!

I'm trying to record in Adobe Audition and it keeps saying my the sample rates for my input and output devices do not match. How do I correct this?

I'm trying to record in Audition and it keeps saying my sample rates for the input and output devices don't match. How do I fix this?

I finally have communication between the ISA One and Audition. I moved the optic cable to another SPDIF on the computer and from the SPDIF to ADAT on the ISA One and it finally agreed that the communication was at the same clock speed. The audio was not intelligible so I moved the optic cable back to the original configuration and I can record voice that is clean.

Audio Sample Rate for Video Clips?

How do you check the audio sample rate for a clip of video?
Thanks.

scroll over in the browser until you get to the column for audio. That's one way.

Project sample rate vs the audio interface clock

I used Audition for digitizing archival audio and I mostly work in 96 kHz because it's the current archival standard for analog audio, however I occasionally need to transfer DAT or ADAT tapes at 44.1 or 48 kHz. One thing I noticed when I went to change the project format was that my audio interface clock did not change and match the multitrack settings. I was able to change the clock manually but I had to go to the Audio Hardware preferences.
I find this behavior for a DAW very strange. If I have a multitrack open of one bit rate and my clock set to another, how does that work? How does Audition 6 play and record audio at differnt sample rates than the interface clock? When I open a multitrack that's 96 kHz and my clock is set to 48, what sample rate am I getting? The audio file being saved says 96 kHz but is it?
Audition 6
RME Fireface UCX
Mac Pro mid-2010

loneraver1 wrote:
If true, this is indeed strange. I have used just about every DAW under the sun for the last 10 years based on what my employer purchased and I have never seen a DAW allow you to record sample rates where the project didn't match the incoming clock.
There should be an option that locks the audio interface internal clock to the opened project. It makes it annoying any time I have to switch between recording in 96 kHz to 48 kHz. That's annoying on it's own right, but what's really annoying is that it's an extra step that I have to train our archival technicians who don't have a techical background in audio.
I can definitely say it's true for my set-up!
Surely, the "extra step" you refer to is not necessary if AA is resampling the audio to match the desired sample rate?
As for the "option" to "lock the audio interface clock to the opened project", for those cards, like mine, which cannot have their sample rate changed by the audio software, how is that going to work? I suspect this was how AA CS5.5 and earlier worked: unless session/project and interface sample rate matched, no audio. I much prefer the current "resampling" method; no necessity to have to think about soundcard/interface settings every time a different sample rate is being used.
FWIW, if I want to record something at 44.1 I always ensure the card and AA settings are 44.1; but for "playback" (or video editing which I do quite a lot of and which requires 48 - my video editor software will not "play" unless the audio sampling rate of the card is 48 - ) 48 is my "default" setting.
JMO!

Changing sample size of PCMA audio sample

Hi All,
I want to transmit the microphone input audio using PCMA codec. I know that the sample rate for PCMA (or g.711) is 8Khz and sample size is 16 bits.
Is there any way I can change the sample size to 8 bits? Appreciate your help.
Thanks in advance.

Hi,
As you can only reset the sample calculation under certain conditions, the system first checks whether the status of the inspection lot will allow you to execute this function.
Prerequisites-
You can only reset the sample calculation for an inspection lot if you have not:
- Marked the inspection lot for cancellation
- Recorded inspection results for the inspection lot
- Recorded defects for the inspection lot
- Made a usage decision for the inspection lot
When you reset a sample calculation, the system:
- Displays information about the activities that have been processed for the inspection lot (for example, sample-drawing instruction was printed, inspection scope for the inspection lot was dynamically modified, physical samples were created, and so on)
- Sets the inspection lot status that existed before the sample size was calculated
You choose Logistics -> Quality management -> Quality inspection -> Inspection lot processing -> Inspection lot -> Reset sample.
You then call up the desired inspection lot and save the data to reset the sample calculation.
Regards,
Narayana.

Sample rate off the audio input and out devices do not match - what to do?

This is fundamental, I know, but nevertheless I can't find my way around it. I get this error message when trying to recor:
Sample rate off the audio input and out devices do not match
and am asked to do this:
Use the appropriate operating system or audio device control panel to adjust the sample rates of the input and output devices to use the same settingt.
I have defined the sample rate to 44.1/16 bit in accordance with my inbuild soundcardt.
I am trying to record from LineIn.
When running on a M-Audio sound card I don't face any problems.
HP 8560W, sound card IDT/High definition audio Codec
Any suggestions?
Knud
Copenhagen

You're sure you have set BOTH the input and output settings to 44.1 16 bit?
Which version of Windows are you running? There are a number of posts on this forum about how to fully access both the Windows Mixer and the Mixer for your soundcard. Especially, you need to ensure that all "Windows Sounds" are turned OFF.

Logic not responding when saving or changing settings & Can't record audio

I have just started to use Logic for the first time (installed it a month ago and now trying to use it).
I have the latest version of Logic, OS X and have a MacBook Pro.
I do not have any external hardware or plug-ins installed.
PROBLEM:
Whenever I try to save or save as a song (even when there is no MIDI or Audio in the tracks) the colored pinwheel spins forever.
I also was trying to figure out how to record audio. I discovered that there was no path for recording audio which might be the problem why I couldn't get it to record (see below), and when I clicked the select path button it also froze on the spinning pinwheel.
Audio Recording Problem:
I set one of the default Audio tracks to Input 1 (built-in microphone) and I see the monitor responding to sounds, but when I turn on the record button on the track and then click record, the mic is still working but nothing is recorded in the track. When I open Garageband and record it works perfectly fine.
Help?

I had many problems with my Mac G5 2.7 refurb/Logic 7.2.1 when I first got it back in March. Although my problems were different than yours some things I remember doing that helped:
Trashing Logic's preferences fixed a situaution where my Logic windows were flashing randomly and Midi info was corrupted causing my Mackie Control Universal and my Korg Triton to freak out Logic.
Repairing disk permissions because 2 times Logic did not totally install correctly. Yes I reinstalled Logic at least 4 times to figure out why I was having freezeups system wide.
Working closely with Apple Care documenting my problems and finally getting my local Apple store in our local Mall to change out the motherboard, dual processors and optical drive. Yes essentially a new computer. Since that time I have not had one system wide freezeup (any wood around here)
or major problem.
AT one point running the "Quick Test" from Apple's startup disc showed a Bad RAM chip. Again not related to you but running the test could show something.
I have 2 issues left to fix - Logic 7.2.1 for me has a transposition problem. Midi tracks will show up after I open a song file transposed down a step. The fix is to reboot Logic which is a pain. 2nd - Annoying beep coming from my power supply. Beep is audible in studio near the tower and is also in the audio chain coming out of speakers. No fix yet but again I'm working with Apple Care.

Changing sampling rate

Hello,
I need to change the sampling rate for a transient response test that I
am running. I am using cFP modules, and LabVIEW 7.1. I am
also using the data logger express VI and all of my channels are routed
to that. If I change my system clock to 250 ms, I can log data
every 250 milliseconds. Likewise for 200, 150, all the way down
to around 30 milliseconds. Below 30 milliseconds, the logger only
logs data approximately every 30 milliseconds. For my transient
response data, I would like to be able to log data every 3-5
milliseconds. Any help is appreciated.
Thanks,
Derek

Derek,
I assume that you are using the FieldPoint express VI. If so, I wrote a simple program to test the execution speed of this VI. By getting a tick count right before and right after calling the FieldPoint express VI, I can calculate the time it took to execute the read. With my tests, the read routinely executes and returns a value in less than 1 ms. It may be that other code in your while loop is causing each iteration to take ~30 ms.
Please try running the attached code, and let us know if you have any trouble.
Regards,
Attachments:
FP test.vi ‏100 KB

Changing sampling time for logging data in DSC

currently I created lots of shared variables by Modbus server. I'm wondering when I active logging data to Citadel, there is any option for change sampling time for logging data or not. I only saw option for changing base on percentage of value differences.
Since I want to export them to MS Excel later, I want decrease the number of sample datas by decreasing sampling time.
Thanks in advance.

This is my idea that i try
First, I add Enqueue Message for Enable time case
Then, I create Enable time cases and write box diagram fallow below pic.
But when i run program,it don't work, time don't count
Please help me.

Changing sample rate during acquisition

I need to programmatically change sample rate during acquisition: My application involves sampling at slow rate most of the time but during "region of interest" I need to change sampling rate to fast (close to max rate of board).

I think it's also possible to use the same (high) sample rate during your
complete acquisition, but in the case you need the 'slow' rate only take
from the array sample 0, 10, 20 etc. or any number you prefer.
In the end I think you want to make sure your computer doesn't have to work
that hard all the way, or if you want to save the data, it's not that big...
If I'm wrong, please tell me.
Ron@ld
Lancer schreef in berichtnieuws
506500000008000000DF570000-1027480788000@exchange.ni.com...
> I need to programmatically change sample rate during acquisition: My
> application involves sampling at slow rate most of the time but during
> "region of interest" I need to change sampling rate to fast (close to
> max rate of board).

On the fence for which sample rate to record at (44.1 vs 96)

Been reading tons of posts on the sampe rate debate. My friends (across the country) and I are about to start to collaborate on the great American rock album that we didn't quite get right back in the day in college. I'll be running the show sending them scratch tracks with clicks so they can lay down individual tracks and I'll import them.
I'm torn on which sample rate(s) to use -- and want the best quality possible, of course. I've boiled it down to the following pros per sample rate. Any advice/comments much appreciated. thanks
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
- Smaller file size to post/share (Dropbox cost per/storage size)
- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
thanks!

rcook349 wrote:
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
- Smaller file size to post/share (Dropbox cost per/storage size)
- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
thanks!
44.1 kHz still is pretty much standard for MP3's.
Your friends/collaborators can pretty much use any application that can record PCM (or even MP3) audio; even if they're not playing to a steady tempo, you can line everything up in Logic, with flex.
Using Garageband and one set tempo should also work. Just remember that you cannot open Logic files in Garageband, only Garageband files in Logic. The Audio Files recorded by either, can be used (imported) by either.
Higher sampling rates will not "future proof" anything. In fact, that whole concept is flawed. Your best bet for now is simply 44.1 kHz 24 bit uncompressed PCM files in their most widely used form: AIFF or WAV.
96 does not noticeably sound better than 44.1, unless you have a top end interface and a very delicate and very complicated mix, and admirably acute hearing. In some interfaces 96 or 88.2 have been found to sound worse than 44.1, because of clocking inaccuracies getting progressively worse at higher sampling frequencies. I would stick to 44.1, it has lots of practical advantages (as you pointed out), and the sonic difference with 96 kHz is marginal at best, and certainly not worth the price: "double" rates need double the CPU power for any plugin processing. That's the biggest loss. Half a Mac.
Bitdepth on the other hand does make a significant difference. There is no reason not to record everything at 24 bits. Shorter: always record at 24 bits.
O, also just spotted your remark about Logic not "letting you" bounce MP3/M4a to 44.1 kHz. You must remember incorrectly, because I never bounce MP3 or AAC to any other frequency than 44.1 kHz. However, it may be that this rate is tied to the projects' sampling frequency as set in the project settings, and the last time I used 48 kHz was in LP 8. I'll check that now.

Changing sample rate corrupted space designer.... somehow.... HELP!

This is a new one to me, so it's very difficult to explain whats happened. I'll start from the beginning.
I created a new, blank logic file with a sample rate of 96.
I created a new software instrument and began to fill it out, adding some compression, some eq, and a little bit of reverb. The reverb came from space designer, using the small room preset "Thicken Vocals". So far so good. No problems, everything's working fine.
I open Reason to rewire in (and ultimately track and record) some drum samples out of reason. The sound is coming out of reason into an aux bus in Logic, and that aux bus is being used as the input for a new audio track. I lay down some midi notes, play back the drum audio out of Reason and hit record on the audio track, essentially making an audio file of the drum samples coming out of logic.
During this, something happens. Rewire is doing a LOT of work balancing the sync, the sample rate, etc.... and somehow, SOMETHING freaks out in the space designer i was using as some subtle reverb for my software instrument in logic (this software instrument is UNRELATED to the Reason audio).
Now, when i hit play, the space designer just produces a loud, annoying pop. If I CHANGE the space designer preset, everything is cool. If i change it back to "Thicken Vocals" under small rooms.... POP.
Worse, it seems to be a permanent problem.
SO --- Ive done a LOT of trying to figure out whats going on. I create a new document and recreate the software instrument. I browse to Thicken Vocals... POP! I tried to replace the IR that "thicken vocals" is referencing by overwriting the file from a system backup I have (an IR called "0.03s_Room Vocals")... still no dice.
Here's where it gets strange. It works FINE if the sample rate is 44.1... I can browse to "Thicken Vocals" and it loads up and init's fine, and everything is golden. As soon as I go to 96, or load it up when the document is set to 96, that's when it freaks out. NOW, if i adjust the falloff curves of the reverb just enough, THEN it suddenly works again, even under 96k.
So, in a nutshell, at 96k, when I load "thicken vocals"... it almost appears (even according to the waveform Space designer draws) to misread the IR waveform, and it plays back as a POP. If I adjust the Falloff curve, it corrects itself, and the waveform even redraws! At 44.1, there are no problems.
I did NOT have this problem before... does anyone have ANY clue what could be tweaking my system right now? I can post more details if I haven't posted enough... including pictures.
THANKS!!
Chris

You might have a corrupted song file. Possibly your autoload as well. Try trashing logic prefs and rebuilding you autoload. Also, repair permissions and maybe even try updating the prebinding in the terminal. Another possibility could be a conflict with the driver of the Audiowerk8, unlikely, but possible.

Sample rate/quicktime pro audio/video

Hi all:
Have a few questions related to Quicktime pro on mac related to sample rate of audio and video...any help understanding much appreciated...
I am a music composer using logic pro and recently started working with filmakers who send me quicktime files of the video to score to...
regarding sample rate of audio as played back in quicktime...I'm a bit confused....
When playing back either an MP3,wave,Aiff at 44.1k or 48k
seems that quicktime plays at the right speed and pitch of audio...Does Quicktime use it's own audio processor to convert sample rates in real time at playback?
When I play a video track with audio recorded at 44.1k or even an MP3...and playback on my soundcard set at 48k...it does play at correct pitch...even though theory of audio tells me that the sample rate is very inportant especially when playing back...sample rate needs to be the same on both the soundcard audio clock and the software...
Is this part of the magic of quicktime or am I missing something.
Even a bit more confused when I got a dialog file from a filmaker attached to his quicktime file that he thougnt was in 44.1k...was actually 48k...so when taken out of quicktime and put in another enviornment for editing that became really important and avident that the samplerate audio rules outside of quicktime applied...
Am I correct then in my understanding that when working only in the quicktime enviornment that the rules of audio sample rate do not apply?
Thanks for thoughts...
regards
vibes

All you may ever want to know about QT, audio and OS X:
http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudio/index.ht ml
QuickTime 7 is a major re-write and it now has many more options for audio. Multiple channel support (up to 24) and sample rates of double DV Stream settings.
Magic is how it works.

Intermitent Pops: Changing sample rate doesn't help

I am capturing to an aluminum powerbook using FCE 2.0.4 from a Canon zr45. I am capturing many hours of footage. I've started noticing every minute or so a loud pop on the audio. It is one frame long and it maxes out the audio levels. Using the suggestion from Apple Help, I switched the audio capture from 48 KHz to 32 KHz (the latter being possibly the original rate of recording). I still have the same intermitent pops on the audio (independent of the sample rate in simple setup). Does anyone have any suggestions of what I might do to fix this problem? Thanks very much in advance!
Al Powerbook Mac OS X (10.4.6)

Thank you very much for responding to my query.
I think your questions have helped, if not solve the problem, at least pinpoint it.
Yes, the pop appears at the same place each time. Yes it is on the quicktime file. And yes, it turns out it is on the original tape! Its faint and not noticeable on the camcorder speaker so I never noticed it before, but when I listened to the tape carefully at the point where it shows up on the FCE file I can hear it faintly on the tape now. I'm guessing it is electronic in nature, as the camcorder was tripod mounted always and the pop lasts for one frame. At least it wasn't a FCE issue. There might well be a way to automatically filter this using an audio filter.
Thanks again!

Changing sample rates for recorded audio?

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