Sample rate of audio!

Where can i find out the sample rate of my audio?
Thank you

Drag the file onto the QuickTime icon in the Dock and hit command-i:

Similar Messages

Sample rate of audio files

I need to view the sample rate of audio files in the info menu on OS X Lion. It appears on Mountain Lion but is there a way to get it to appear on Lion?

Drag the file onto the QuickTime icon in the Dock and hit command-i:

Sample rate and audio-MIDI sync issues

Disclaimer: I did read other posts similar to this but couldn't find an answer to my specific situation. So here it is:
Logic was perfectly fine when everything was running at 44.1kHz sample rate. Then I got vocals at 48 kHz so I had to convert the sample rate in Logic Pro to match it.
Suddenly I get a slew of "Error trying to sync MIDI and audio" messages. After crying and changing the sample rate back to 44.1, then to 48, and over and over again, it finally works again.
So Logic is fine at 48kHz. But when I go to a track that's at 44.1kHz, I get the sync messages again and have to play "toggle the sample rate" for about 5-15 minutes before Logic decides whose master again.
Why is it doing this? Do I need to change Logic Pro to some kind of default settings every time I go from one song to another with a different sample rate? Or will this not be an issue if I upgrade to 7.2? (I have 7.1.0)
PowerPC G5 Mac OS X (10.4.6)

This is not a bug, but a nuisance.
You should upgrade to LP 7.1.1, which is way more stable than 7.1.0. No need for you to go to LP 7.2.
"Then I got vocals at 48 kHz so I had to convert the sample rate in Logic Pro to match it."
Are you certain, that in your song, in your regions, you used the newly converted 44.1kHz files, chosen from the Audio Window, and not (still) the old 48kHz files?
"So Logic is fine at 48kHz. But when I go to a track that's at 44.1kHz, I get the sync messages again"
LP doesn't do this well. And for a reason, but we'll not get into this now.
See: "Audio > Sample Rate > ..." and select one.
Perform proper conversion and make sure ALL of your audio files running in your song are congruent. Check your Audio Window and the files associated to the regions.
Set up your autoload to contain these settings. From then on, whenever you know that you will be importing other sample rates, change the settings in "Audio > Sample Rate > ..." before loading the sounds/files, if indeed you are starting from scratch. This will save you significant time.
Been there, done that.
sonther

Goofed my sample rates, now audio won't sync 44.1kH versus 48kH

We recorded the audio for our project using a Zoom box that had the sample rate set at 44.1kH. I opened the files in Soundtrack and diced them up, exported them and then processed them in iZotope's RX.
They are now 48kHz, and of course the conversion from 44.1 has thrown off the speed of things. Can't Synchronize Clips in Final Cut.
Is there any quick fix for this? I don't want to have to go back to the beginning since the editing and processing steps mentioned above were very time consuming.

Goldfish wrote:
Charles Floading wrote:
I opened the files in Soundtrack and diced them up, exported them and then processed them in iZotope's RX.
They are now 48kHz,
When and how did they become 48K?
My guess is that at least some of your work will need to be re-done depending on when you did the conversion to 48. If you did all PRE-video audio work at 44.1 and then exported the result as 48 you should be ok.
They were 44.1 going in to Soundtrack. When Sound track exported the clips, they were saved as 48k .aiff files. From there I went into RX, and then to Final Cut.
Over the course of a 15 minute clip, the drift isn't huge, but it seems perceptible. Fortunatly there are no shots that last that long until cutting away to another camera angle. I guess that worst case scenerio, I can manually nudge things if need be.
Thanks for the help so far everone. Any other suggestions are welcome.

Determining bit and sample rate of audio file.

Bit of a newbie, and i'm wondering how to determine what the sample rate and bit rate that an audio file has been previously recorded. I'm using Mbox 2 Pro.
Thanks

Open the Audio window and import the file.
cheers
rob

Finder - audio file info - bitrate, sample rate, etc

Is there a way to find out what bit rate and sample rate an audio file is without opening it in QT and then CMD+I from there?
I don't see detailed info when I get information from Finder. Is a script or anything that will allow me to find this info quickly in Finder?

juhani h. wrote:
Works fine here, too. But make sure you´ve ticked the box "Convert audiofile...etc", like M-Dan pointed out.
I tested it and if unchecked the imported file plays at the wrong speed. Amusing at first, but potentionally disastrous.
/J
Does NOT work in repeated tests on my system, and yes, the 'convert audiofile' box is ticked. Wouldn't report it as a bug if it wasn't....
This almost ruined a session of mine and cost me a deadline, but luckily I caught the SNAFU in time and converted the files individually.
Very odd though to hear that it works on some users' systems. How can this be a system specific bug? Are you sure you're dragging files in from the Finder to your arrange page? And Logic converts them? Are we talking 48k files being converted to 44? Or vice versa (44 to 48)?
In my case, it was a 48k session and Logic failed to convert the 44k files I had mistakenly been sent.
Since I've noticed other weirdness (like malfunctioning Apple Loops) at 48k, I still think Logic is less than bullet proof at different sampling rates. Just a hunch though.
T

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.

Problems with audio sample rate when capturing DV50

Hello from Spain, I have a problem when capturing DV50 source footage: I setup my sequence settings and capturing settings like always, DV50 codec and 48k sample rate for audio, then when the capturing process finish the result is a file that contains audio to a non-48k audio sample rate, then video and audio don´t match. The sample rate is 47945 and other similars, not always the same, sometimes 47214 sometimes 48000, and there is no rules...My capturing device works ok in other computers and the footage has been capture ok in other Mac...I don´t know what happens.
Sebas, kind regards
G5 Quad Mac OS X (10.4.6)

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...

Audio sample rate 192 k but seem less than that (about 20% less) when i run software defined radio .

I bought Pavilion dm4-1000au with Win7 Starter as original OS. My target is to get full Sample rate of Audio 192 K when I ran Sofware Defined Radio like Softrock SDR. Actualy I only get less than 100% of sample rate. So, frequency coverage of SDR become less than I prefered 192 KHz.
What I have done is up dating Audio driver from HP Support/ Driver Download center only.
How to solve this problem ?.

Thank you for your input.
YES - I've seen the "right click Time Code display and switch to audio units timeline solution - yet audio units do NOT appear as an option for me...hence all of this drama.
ADOBE - do I have a munged install? Any other advice? We keep a clean system (Mac Pro 10.6.11 or whatever is last iteration of SL). I guess I could try installing into a new User and see if it behaves any differently.
Sigh...we bailed on FCP7 after trying FCPX and also passed on going to Avid MC. Hope PP is reliable for our work flow...again thx for all help.
PCG

Changing audio sample rate in FCP

Hi all
A quick question, can someone explain how to change the sample rate of audio in FCP, if at all possible. Or do I need to do it in Soundtrack?
Many Thanks
Adrian

You can't change the sample rate in FCP; you can only change it by exporting it. You can do that from FCP, STP, Compressor, QuickTime, iTunes, and a bunch of other applications. Making a Compressor droplet to do this is perhaps the easiest way.

Audio samples rate

Hello ,
Can someone tell me wht are the samples rate of audio files suported in Flex ?
I try to play wav files, and i tried with 44100 hz and it is ok, but another file that have 8000 is not played good.

Maybe i wasn't realy explicit in what i asked.
I used library as3wavsound to play wav audio files. And i noticed that files with samples 11025,22050,44100 are played ok, but 8000 and 16000 not. Also i found an article where is written that Flex supports only the frequences wrote above, but i couldn't find this information on Adobe documentation.
Maybe someone knows if it is possible to play a wav file at fquency of 8 and 16 kHz and some examples maybe??

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!

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.

EMU 0404 USB - DS driver and sample rate autoswitchng

It looks like autoswitching works well until output is going through the ASIO driver. With output switched to DS the "sample rate" option in control panel becomes grey and do not change its value to reflect source sample rate.
Assume nothing is playing via e-mu then I can switch Fs manually in audio control panel to say 96 kHz. Then I start playing some 44.1 wav via foobar + DS out. "sample rate" in audio control panel doesn't reflect 44.1 and remains greyed with Fs = 96.
Is it feature or bug?
What is going out from SPDIF in such case: bit perfect 44.1 or 96 with SRC?

Hmm... there are plenty of possible reasons for this ...here are couple of them (not mentioned in your list of suggestions given by Creative support)
- multicore support could be one culprit (I don't know if the FL has this option but if you use (IIRC) Cubase 3-> )
- on Cubase, another possible culprit is an option named as Lower latency ... disable this feature
- some background process starts after some period (E-MU USB is quite sensitive for these) ... run the DPC Latency Checker utility (just google the name to get it downloaded)
Does Coscos Reaper (google the name) give same issue?
jutapa

Sample rate not conforming to project

I have the box checked in the settings that is supposed to conform the sample rates of audio regions to the project, but it is not converting.

You might find this discussion useful: https://discussions.apple.com/thread/5114708?start=0&tstart=0

Sample rate of audio!

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