Sample rate change from 44.100 to 48.000

Hi Guys,
I am preparing a sound design file for a film project. the audio files are originally in 44.100. Now, I am preparing the surround mix for the DVD release. For DVD I would need a sample rate of 48.000khz. However, when I change the sample rate from 44.100 to 48.000 in the Project settings, all my audio files seem to be shortened in the arrange and everything sounds as if it was sped up. How can I change the rate to 48000 without the audio files changing?
thanks

Man you have to convert the files to 48khz otherwise they will of course sound mickey moused and distorted when played in a 48khz project. You can easily do that in the Bin, batch process all your files and convert them to 48 using the "copy and convert" funtion from the menue. Compressor does the job also...

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So who can tell me why my project in which all the peices are saved 41k are magically transfered to 48k when saved as an MP3?
Cheers.

Because your MP3 encoder isn't set up properly? Have you checked all the options available when you save as .mp3?

Bit rate changes from 16 to 32 when resampling...why?

Hello,
I have a few DV Pal Quicktime Movies which have an odd sample rate - slightly above or below 48Khz - this is a problem with the camera which I know about.
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This is very interesting isn't it !
I have had a similar problem in porting audio between STP & Pyramix, a PC based DAW - often files that I have exported from STP will be recognised as 32 bit files in Pyramix, resulting in gross distortion, and a totally unusable file. This is most frustrating.
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b) an effect file (if you've got some effect on it, or have spliced it etc)
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Hello,
So it seems that changing the sampling frequency is affecting the gain of a channel?
I am not that sure what is causing this behavior, if you could answer the following questions, we could begin to figure this one out.
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3.) Do you get this error if you call DAQ_Clear before reconfiguring the sampling rate?
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Jus
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Hi
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Audio sample rate change

Hi; am importing audio into FCPX - AAC, 44k1 - along with the .mov vids, but on (trying to) export finished work am finding it nearly impossible to alter the 48kHz (default?) output sample rate .. how so ?
Any help gratefully received .. cheers
Peter

Why don't you want the 48kHz?
It's the standard for video.
This article may help (audio is at the bottom).
http://support.apple.com/kb/PH12526?viewlocale=en_US
Message was edited by: Ian R. Brown

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Hi
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I have a problem that exists on serveral levels. Hopefully I can explain:
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I have a problem that exists on serveral levels. Hopefully I can explain:
1. I directed a concert shoot with eight cameras. All cameras were Panasonic DVX100A configured for 24p Advanced except one, which was a Panasonic SDX900 configured for 24p (which I later discovered was laid down with a 3:2 pulldown instead of a 2:3:3:2 advanced pulldown like the DVX100A cameras).
2. It was planned from the beginning to use the multiclip feature in FCP to edit the footage.
3. The tapes from each camera were captured from beginning to end—about an hour each tape, four tapes per camera—at full resolution, the pulldown was removed, and the clips were then recompressed to Photo JPEG EXCEPT the tapes from the SDX900.
4. When I finally figured out that the footage from the SDX900 used a 3:2 pulldown, I opened the captured clips in Cinema Tools and attempted to remove the pulldown. The problem I ran into there was that CT can't handle files larger than 9 GB. Each captured clip (four of them—one for each tape) ranged between 10 and 13 GB.
5. My solution to this was to open each clip in QuickTime Pro and split the clip in half, open each half in CT and remove the pulldown on the smaller files, then open the smaller halves in QT again and join them back together into a single clip.
6. When I open the new master clips in Cinema Tools and analyze them, the video/frame rate is 23.98 as expected. BUT, when I import the clips into Final Cut Pro, the video/frame rate is 29.97 and I can't understand why. If I try to do a pulldown removal (again) from FCP, it doesn't work—I get an error that it can't find a clip with a pulldown to remove (duh!).
Is this a bug, and is there any way to fix this and make it work? I need to make all of the assets available for my class at school.

SCXI Sample rate changes when VI run as SubVI

I have a VI which acquires 7 channels of data from SCXI-1100 module and places the data into a global variable. The VI runs fine on it's own but once it is used as a SubVI the performance drops.

Hello Andy,
Using global variables might be the culprit here. You have to consider that for every read of a local or global variable in a VI or its subVIs, a copy of the data is made in memory. Each instance of a local or global variable will therefore make a copy of the data in memory.
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Does Reading a Local and Global Variable Create a Copy of the Data in Memory?
http://digital.ni.com/public.nsf/websearch/37002ACC84B961CA86256D9C00760EE2?OpenDocument
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Todd D.
NI Applications Engineer

Automating wav import, sample rate change, mp3 export

I do radio spots, lots of them, and every day I have to take my spot that is a 48k wav, convert it to 44.1k, then convert it to a mp3, them distribute via email.
Can I automate this task?

Rather than trying to automate iTunes, it might be easier to do the conversion in one go with a utility such as Sound Converter or Switch.
Hope this helps.

IPhone: AudioQueue - is it possible to change the sample rate?

I've been playing around with the AudioQueue stuff for a few days and it's all working fine.
I was trying to build a low-latency playback system by making the streaming buffers the same size as the audio file and pre-loading the buffers (which works fine) but I've hit a snag.
I've been trying to get the streaming to work at different sample rates so that I can play back the same sample at different pitches. I managed to do it by modifying the sample rate in the AudioStreamBasicDescription structure but in order to actually make the stream playback at the new rate it seems you have to create a new output, reload the audio file into the buffers and re-enqueue the output queue before starting playback again, otherwise the sample rate change has no effect.
There is a method to set queue properties; AudoQueueSetProperty() but unfortunately the sample rate Property (kAudioQueueDeviceProperty_SampleRate) is read-only
Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
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Dan,
there is one point in your understanding, which i am not sure what you think about when talking about it: I understand E-series devices do not support this property change while the VI is running.
infact, you cannot change the sample clock rate during acquisition. but
this does not mean that you cannot change it while the VI is running.
you have only to interrupt the acquisition. since you want to acquire
continuous, this would have the same effect as stopping the vi, i asume.
so the best way to accomplish this task is to use an external clock.
this is e.g. often used for acquistion on rotating shafts. the
acquistionrate is always e.g. 24 points per revolution regardless of
the rotational speed of the shaft, except for a maximum frequency of
course.
Norbert B.
NI - Germany
Message Edited by Norbert B on 09-14-2005 04:16 AM
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

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

About Logic's Sample Rate

When you choose a sample rate, I know that the sample rate will be applied to the incoming audio of your interface BUT what if you have a project of music that contains Logic software instruments? Does the sample rate change the audio coming from the software instruments also?

Hi,
I don't know, but I think this only applies to software instruments when they are a) frozen b) bounced. Because: The things you hear coming out of you monitors cannot have a sample rate - they are analog 20-20,000 Hz. Period. The question is which way Logic takes to create this analog signal. It's a question how the software instruments themselfes create the signal. So, when sample rate is 44,1kHz, do the software instruments "run" their output at a lower rate, too? Or do they always run at full pace (192)? Or always at 44,1kHz, since this is way enough for the demands of a synth?
For EXS, the sample rate is determined by the EXS samples. See how much they have: That's what Logic will feed into the D/A's.
Don't know how the synths are running, though.
Fox

Sample rate change from 44.100 to 48.000

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