Sample Rate Conversion in Sample Edit Window

The option to sample rate convert in the sample edit window appears to have disappeared (was in Logic 7) This enabled destructive editing (sample rate conversion) of a selection of an audio file. There are a number of sound design or problem fixing reasons I have used this feature in the past.
The only option is now to copy, convert and replace the whole file and as this is not destructive if you wanted to apply the same changes to a number of different mixes (saved as different songs) this is now also no longer possible.
Is this feature available on another menu? (Time and Pitch machine will not do the same function)
Julian

Hi tslr...
I am thinking that ever since Apple came out with Core Audio architecture they basically put it in every audio app that they had, that said- Apple's buyout of Emagic was most probably where they were able to get allot of the Core Audio architecture from, (there is a memo somewhere about this- quite public knowledge)...
Basically after OS9 and the buyout it was really cool to see parts of logic running around the OS, (Midi preferences etc.)
Then Garageband...
I think what I am trying to say is that this is most probably some functionality that is occuring due to Core Audio Services- not Waveburner- so most probably there is a release note somewhere discussing Sampling Rate Agnostic's. (Actually come to think about it I think it was a new feature of Logic Pro...
Regards
Alex
PS: I wouldn't think that it would matter about being in 48 or 44.1- so don't loose sleep over it.

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Apogee 16X, Gigas, Sample Rate Conversion, and Summing/Monitoring

I'm trying to do an update on my home studio rig.
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Dual 2Ghz G5 Mac OS X (10.4.7)

96k (at least) is pretty important, because as well as doing the normal sort of workaday film and pop level audio, I'm also doing this very intimate project for an audiophile vinyl company, and it's stipulated that I must use 96 at the very least. I'll probably be doing that stuff at 192, and I will be keeping almost everything third-party off the processor. The company would prefer that I did everything to a 2" analog 8-track, but that's where I drew the line. Thing has to be aligned every three hours of operation, cause the track widths are so high, and the thing's pretty old, I'm afraid.
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Real time sample rate conversion?

Recently upgraded all my hardware and software.. I want to record audio at 96khz - so I set my hardware to 96 and set logic to 96 in audio - no problem... except when I want to record new audio on an old project - I set the software to 96khz, and hey presto my old files recorded at 44.1 play too fast....
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If I have my hardware set to 96khz but Logic to 44.1 so the old audio plays okay will my new audio still be recorded at 96?
I could probably convert all the files individually using the sample rate converter in Factory, but I want to avoid this lengthy process as I'm talking the last 8 years of work!
Anyone out there know anything about this?

Just fancied recording at the best quality possible - but you are right, the difference won't be that noticable and may even sound out of place.... in any case I can record at 24 bit with no adverse effect...
I noticed that trying to record audio at 96 when my hardware is 96 and logic at 44.1 has the effect of serious latency issues plus disk too slow errors. So forget that one.
Someone told me recently that the difference in bit resolution is more audibly noticable as a change in audio quality than sample rate... just out of interest I wonder if anyone knows if this is true?
BTW just recording guitars and stuff, some vocals, band kinda stuff...

Sample Rate Conversion of ALL tracks at once...

Hello everyone. I've got a bit of a problem here...
I've tracked a whole song in 96k 24 bit, about 20 tracks - big mistake! For whatever reason, I get about 20% of my UAD cards used and 60% cpu usage and Logic just can't play the song, I get that "can't be processed" message. So, either my mix is going to be lame and way too raw, or I have to convert all files in the song to 44.1K. I've done many other projects in 44.1 and I get hundreds of plugins before Logic and the G5 come close to overloading.
I know I can use the Audio Window to convert the files and save them in a new audio folder, but how can I reconstruct the song? I can place the files into a new song at 44.1 and eyeball where to put them, but if one file is off by even 1 sample I get major phase issues.
Is there a way for Logic to convert all the files at once, and place them each in the exact same way as the 96k version? Kind of like a batch conversion for the entire song and arrangement.
Thanks a bunch

Oh, I think I figured it out. I converted all of the files, saved them in a new folder, changed the sample rate of the song, saved it, closed Logic, replaced the 96k files with the 44.1 files of the same name, reopened Logic and the song, said ok when it said the files had changed, created the overviews and bingo, the same song at 44.1, plugins, automation and all!

Core audio and sample rate conversion

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.
The annoyance is when one forgets to keep track of core audio and inadvertently ends up listen to a piece of music sample rate destructed. You know - walk up to music server computer, forget that you had DAC set to 44.1k for last music played, put on 96k source, SRC takes over and you don't notice the artifacts and distortion for a few songs. No one wants that! Don't get me wrong - it's a convenient feature and the amateur user would be sunk without it.
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?
Thanks

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.

Sample rate conversion results in chipmunk voice

When I convert the sample rate from 44100 to 22050, I sound like a chipmunk. Why?
Power MAC G5 Mac OS X (10.4.7)

My reason - I do voiceovers for a living and I have a client who wants his .wav files delivered at 22K. He inserts them into flash for on-line automotive training.
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Power MAC G5 Mac OS X (10.4.7)

Am I missing something - automatic sample rate conversion?

SETTING
Logic Express Version: 9.1.6
Project Sample Rate: 96kbps (shown in the Transport Bar)
File > Project Settings > Assets tab: "Convert audio file sample rate when importing" option is selected
LIGHTS, CAMERA, ACTION!
When I importa a file via File > Import Audio File or dragging & dropping an audio file the sample rate is converted to 44.1kbps.
Is this a known bug, or will I need to dig deeper to find the error in my ways?
cheers

ah ha - eye, brain & fingers not all in sync. that is 44.1 kHz.

Audio sample rate conversion

Hello again.
In my timeline I have audio on track 1 from a mini DV - sample rate 32kHz.
On tracks 3&4 is a stereo track from a CD at 44.1kHz.
If I try and play together them the audio on track 1 is mute. However if I disable the monitoring on 3&4 and then play the timeline I can hear track 1 fine.
I assume its a sample rate issue though I'm probably wrong.
Can anybody shed any light on this.
Many thanks

Use Soundtrack to convert all your non-compliant audio to 16 Bit 48kHz AIFF files.
Your problems will disappear.
Set up your camera to shoot 16 Bit 48kHz for next time.

Sample rate conversion help

Where do I set Logic to convert imported audio files automatically? I'm confused because I (think I) often drag 44k files into 48k sessions and they seems to be the right pitch (could this be wrong?)
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I know in PT, I can drag any sample rate into a session and it will automatically be converted to the correct rate (in other words converted so that it plays at the correct pitch)
Is it possible that Logic does not do this?

I don't know - when I need to convert files (usually from a PC) into an existing session, I usually use an external editor to convert them so I can preview them before importing in to Logic. I know Logic converts samples on the fly, but I'm not sure about audio files, sorry.

Max sampling rates in differential sampling

I am rather a novice in terms of DAQ and wonder about the maximum
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Hello Sirnell!
You will have the same sampling rate regardless of which connections you make (differential,RSE, NRSE), but with the differential connection you will reduce the amount of channel you can use. With an E-series board with 16 inputs you will only be able to connect 8 signals using the differential connection.
For more information about Field wiring take a peek at this link:
http://zone.ni.com/devzone/conceptd.nsf/webmain/01F147E156A1BE15862568650057DF15
For more information about DAQ (glossary):
http://zone.ni.com/devzone/conceptd.nsf/webmain/45ACC30D4A769A3F862568690061D750
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Ashwani S.
Applications Engineer
National Instruments Sweden

How to improve sampling rates in mega samples spartan 3e board

I am trying to acquire data in spartan 3e board in mega samples rate can you post me any examples for this purpose
Solved!
Go to Solution.

If you're trying to get analog samples in that range, the issue that you're going to run into is that that sort of sampling rate is right at the bleeding edge of what you're asking for (it's a 2-channel @ 1.5Msps chip, the LTC1407A-1). If you actually want to capture at these rates, you're going to need to pull some interesting LV-foo using single-cycle timed loops in the FPGA diagram or drop some code down into an IPNode/CLIP Node, with outputs from those IP blocks hooked to the IO in a SCTL.
Please let us know a bit more about your specific design.
-Brad

Sample rate for digital sampling (cDAQ-9172 & NI 9401)

Hi!
I have a cDAQ-9172 with a NI 9401 C-series module (digital). I would like to sample the digital inputs with a sample rate of e.g. 400 kHz or 200 kHz. My problem is that I can only select a the 100kHzTimebase clock, and therefore only get a 100 kHz sample rate. The 20MHzTimebase clock is too fast, since it gives me a sample rate of 20 MHz). Is it possible to get a user defined sample rate of e.g. 200 kHz, by e.g. dividing down the 20MHzTimebase clock?
Solved!
Go to Solution.

The cDAQ-9172 chassis does not have an internal timing engine for digital input however you can use one of the onboard counters to generate your clock. Set your pulse train generation counter to be one of the internal counters, such as "cDAQ1/_ctr0" and your digital input sample clock source to be /cDAQ1/Ctr0InternalOutput".

JavaSound+sample rate Conversion

hello all
I have a small problem for the conversion of a file with audio.wav (8kHz, 16.1, true, false) format
to a audio. wav but this time with (16KHz, 16.1, True , true) is that this is possible with Javasound?? and how I can do it??
pleeeeeeeeeease help
thank's in advance

thank you for replay,
but it's not working so what I'm doing wrong, this the code:
and thanks in advance:
import java.io.File;
import java.io.IOException;
import javax.sound.sampled.AudioFileFormat;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.UnsupportedAudioFileException;
* @author Jinio
public class SampleRateConversion {
     public SampleRateConversion(String myFile_wav) {
          // for exemple: myFile_wav="c:/song.wav";
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          AudioFormat format=null;
          // TODO Auto-generated constructor stub
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                    nWrittenBytes = AudioSystem.write(stream, AudioFileFormat.Type.WAVE, outputFile);
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public static void main(String args[]){
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Can I automate import, then sample rate conversion, then 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.

Sample rate conversion comparisons

Seen this?
http://src.infinitewave.ca/
My apologies if this has already been posted here.

It's been posted before (more than once) but it's worth reminding people, especially as it's been updated. A few more have caught up now, but it's interesting that software like ProTools and Pyramix (which costs a fortune) still manage noticably iferior results compared to Audition - and the latest Audition results are even better than the previous ones. Many of the ones that look okay on a sine sweep fall down quite badly because they cheat, and don't convert right up to the Nyquist point - as the frequency graphs demonstrate. The only conversion that is as good (indeed slightly better!) across the board is the iZotope RX2 advanced High Steepness one, but perhaps that's hardly surprising... and it costs a lot. For all practical purposes, the Audition conversion is the best overall, because it's also good value for money.

Sample Rate Conversion in Sample Edit Window

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