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....
The ref manual says logic will do real time sample rate conversion, but doesn't suggest how this is done.
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...

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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.

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

I'm trying to do an update on my home studio rig.
I've decided to get the new Quad Core 3g G5.
I've decided on the Apogee 16X/Symphony card combo.
I need to rout the outputs of six PCs into Logic; One PC is running KYMA/Capybara, one PC will be running the Native Instruments Kore/Komplete VST host; and the remaining four are running Gigastudio 3. All the PCs are lightpipe out. At present, I run the lightpipes into a Hammerfall lightpipe-to-MADI converter, and from there onto a MADI card via coax directly into my present G5, the Dual 2g.
I monitor though a Sony DMX-R100.
So far, no problem.
But, I've decided to start working at 96k, so my present scenario becomes more difficult, as my sample library (about 1.2 terrabytes) is all 48k.
I'd sort of like to get rid of the DMX, but Im considering using it as a submixer for the PCs, running analog out of it into 12 channels one of the Apogees. That would solve the problem of sample rate conversion. But it would still leave me with a pretty big piece of gear that I'm not sure I need.
The other thing I'm wondering is about summing and monitoring... I'd like to be able to avoid ganging everything rhough the Logic 2-bus, I like it better when I can spread things about a bit in groups, also there are a few other things (like the movie audio and a DVD player) that I need to be able to monitor through the same system.
I also would like to be able to use my Fairchilds and Pultecs on the 2-bus of whatever I'm monitoring through, which would mean somehow returning that through Logic, or using my 2g Dual as a sort of mix&stem storage/archive/networking computer. (Which I'm not wholly opposed to.)
Any ideas?
Dual 2Ghz G5 Mac OS X (10.4.7)
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.
For the film and pop stuff, I still like 96k, cause... I dunno why, I guess, now that you mention it. But even when stuff is downsampled, I still think it sounds better when its recorded at higher resolution.

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.

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.

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.

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.
How am I converting - In the Audio Window, Factory, Convert. I type in the convert rate I want.
BTW, I was told by a Logic Pro phone tech that this would work.
Any ideas?
Power MAC G5 Mac OS X (10.4.7)

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?)
Now i'm dragging a 48k file into a 44k session and it's playing slow. I tried converting it manully but got an error. I read a workaround on the forum about changing the session rate then bouncing a file at the new desired rate, but that seems like more work than should be necessary.
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.

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";
          File inputFile=new File(myFile_wav);
          File outputFile = new File("c:/test1.wav");
          AudioInputStream stream=null;;
          AudioFormat format=null;
          // TODO Auto-generated constructor stub
          try {
               stream=AudioSystem.getAudioInputStream(inputFile);
               stream = convertSampleRate(16000, stream);
               int     nWrittenBytes = 0;
                    nWrittenBytes = AudioSystem.write(stream, AudioFileFormat.Type.WAVE, outputFile);
          } catch (UnsupportedAudioFileException e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
          } catch (IOException e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
     public AudioInputStream convertSampleRate(float fSampleRate,AudioInputStream sourceStream)
               AudioFormat sourceFormat = sourceStream.getFormat();
               AudioFormat targetFormat = new AudioFormat(
                    sourceFormat.getEncoding(),fSampleRate,sourceFormat.getSampleSizeInBits(),
                    sourceFormat.getChannels(),sourceFormat.getFrameSize(),fSampleRate,sourceFormat.isBigEndian());
               return AudioSystem.getAudioInputStream(targetFormat,
                                                            sourceStream);
public static void main(String args[]){
     new SampleRateConversion("src/test.wav");
}

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

Hi Guys,
Is there a way to convert an entire Logic session from 44100 to 48000 without the audio files warping?

You'll have to convert all 44.1 kHz files in the audio bin to 48 kHz.

Film guys: Sample Rates?

Do people work at 48khz in Logic, or work in 44.1 then change rates later?
I'm working in 48khz just now, but it means that importing 44.1khz files is a drag, seems they need to be converted to playback properly. Working at 48 also seems to mean that my 828MkII is 'stuck' in 48khz, so that neither iTunes or Waveburner will playback as long as Logic is running.
Is there a more elegant solution, or just "shut-up and get on with it"? ;o)

Irv,
I'm surprised to hear that you can't get iTunes to playback while Logic is running @ 48KHz. I never have a problem with this. Perhaps it's got something to do with my audio system, tho I'm not sure. One thing I could swear I remember reading is that CoreAudio will do real-time sample rate conversion, which (I guess) explains why I can play an audio CD while my session is @ 48K.
I just did a project where most of the music was originally recorded at 44.1, the mixes and stems ultimately bumped up to 48K. Personally, I'll never do that again if I can help it, for a variety of reasons. One, because it generated too many files with similar names (strings stem 44, strings stem 48, etc.), and Two, because when we had to do tweaks, we had to go back to the 44.1 session, re-print stuff, then redo the sample rate conversions. Real PITA.

Help please? Logic playing back wrong sample rate

So I created a new session and recorded a new song that I have already done some work to this song and have had no problems whatsoever. Last night I went to open this song and work on it some more and when I pressed play the song started playing back very slowly. This session was recorded at 88.2k and I am getting a Logic error message that says sample rate 44.1k is recognized however my audio interface (a Metric Halo 2882 +DSP 2d unit) is set correctly to 88.2k. I look at my bin and all the files are correct, my preferences are all correctly set at 88.2k but Logic is playing the song at about half speed!!
What happened?? I have a couple projects going and I need to get this fixed.. Anyone??
Thank you in advance!!
DDD
Mac Book Pro Dual Core, OSX 10.6.8, Logic 9.1.5 (do not want to upgrade Logic to 9.1.6 I tried that and it was a BAD experience so Im sticking with 9.1.5), Metric Halo MIO Console 5.4

Sounds to me that your best bet is to use a utility like SoundHack (http://music.ucsd.edu/~tre/soft//SH893.hqx) to change the audio file headers. What you want is to flag the files as 48k, not do any sample rate conversion--they should play back natively at 48k. The only SRC you'll need to do is if you actually do want the files to end up as 44.1k at some point. (In your case, I don't think Logic is converting the sample rate in real time, by the way--what it's doing is playing back the files with a clock setting of 48k, which only sounds right because they've been mis-labeled.)
Either way, you should fix the file headers now, I think. And even if your final product needs to be 44.1k, since your files are quite clearly at 48k now, I'd do all the necessary editing and processing at 48k, then convert to 44.1k as the final step. And until the SRC in Logic is improved (Apple ?), maybe use some other app to convert to 44.1k in the end. Peak gets high marks--not sure what else out there sounds good and doesn't cost a mint.
Best luck!
James
[email protected]

Real time sample rate conversion?

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