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.

Similar Messages

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.

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...

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.

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.

Green & Pink horizontal Lines? and Audio Sample Rate? Help, Please!

I have just successfully completed one short film. I'm believe I'm doing exactly the same as before but I'm running into problems. Used a Panasonic PV-GS300 camcorder with mini DV. Then a Sony DSR-11 deck. When I capture there are green and pink bars within horizontal lines over the image in the viewer. When I end the capture I get the message 'the audio sample rate of captured media files does not match the sample rate on you source tape. Video and audio out of sync.' The captured film clip audio rate is 48.0 KHz, the audio format is 16-bit integer, the audio is 2 Mono and the tracks are 1V 2A. Any suggestions, help would be greatly appreciated.
1.8 GHz Power PC G5, 2GB DDR SD RAM Mac OS X (10.4.9) FCP 5.1.4

Sorry about that, I'm away from my FC system, so I can't refer to it. You should choose DV-NTSC in Easy Setup as a starting point, then if the audio settings of your footage are different from 48kHz 16-bit, then you can easily change the capture settings to match.
tim

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)

Geting audio sample rate error, help

Hey all, been doing a massive project where I ma bring in tons of old 8 mm tapes, hi8 and digital 8. This one tape I brought on though however is giving me grief. I keep getting this error:
The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape.
If find this odd though cause the tape should be standard, NTSC dv 48 K. Any suggestions. Also how do I reset my final cut so that when I plug in a camera it always reads it as what it recognizes. I ask this because I had to used the setting uncontrollable device because on the original old 8mm tapes there is no time code so I had to capture that way.
Anyhow, any suggestions on this would be great for if I recapture. Cause I could line it up by eye but want to find the problem so I know for the future. Thank you.
Nathan

This is a recent problem that seems related to a recent upgrade of QuickTime. Here's why.
In the last month, a rash of these posts have begun to appear:
"DV Capture Audio problem"
http://discussions.apple.com/thread.jspa?messageID=6708693&#6708693
"audio/video"
http://discussions.apple.com/thread.jspa?messageID=6591262&#6591262
Plus this thread, plus my own.
In my case, nothing changed in my operating system or Final Cut Pro version. I upgraded to QuickTime 7.1.6, and the problem began. I have upgraded all the way to 7.4 to no avail. When I attempt to import a DV clip using the same Sony DVCAM deck that imported the same clip in December, I get the error every time. Nothing has changed in the tape, the deck, the Project or Final Cut. I am simply unable to import DV video. I can import other kinds (Panasonic P2, for example), but DV is a no go. I cannot get rid of this error.

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.

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.

Frame rate conversion HELP

Is it true there is no way to SUCCESSFULLY apply pulldown to 24pNative footage (shot with the HVX-200 on P2 cards) OTHER than laying it to tape? I'm told it's the video card that's doing the pulldown and there's no way pulldown can be done within FCP/Cinema Tools/Compressor. What I'm trying to do is convert my DVCPRO 50 23.98 quicktimes to DVCPRO 50 29.97
I just can't come to terms with the idea that the only way I can work in a 29.97 timeline is by dumping all my 24pNative footage to tape and re-digitizing. Of course you can drop a 24pNative clip into a 29.97 timeline and render it and FCP will "pull it down" so to speak by adding duplicate frames but of course it isn't correct, nor does it look correct.

Hey Shane,
Thanks for all the help so far, very much appreciated.
THE WHOLE STORY with a happy happy ending
- shot a music vid with the HVX-200 and the mini-35 adaptor (with SLR lenses) in DVCPRO HD 720pN mode. Planned to deliver the video in standard def.
- Chose HD 720pN recording because a) wanted to be able to have the option to deliver an HD version if ever desired b) wanted to be able to use the HVX's undercranking and overcranking abilities only available in 720p mode (which look amazing and can be played back on set real time, I'm still to this day impressed by that), 3x the recording time on a P2 card at 720p 24pN versus recording DVCPRO 50 24p, and finally wanted my VFX/compositor guy to be able to work with highest res images possible (i.e. do his work with the HD dailies and deliver me HD and SD versions of the completed VFX shots ). And yep, in order to deliver an HD version of the video I would have to do a match back edit with my HD dailies and HD VFX shots but that really doesn't take too long for a 180 second show.
- Chose to work in and deliver SD because it's all my offline rig is capable of outputting (i.e. no HD client monitor, just my NTSC CRT). I know some people who cut their music vids (shot in DVCPRO HD 24p 720p/1080i who are still going to deliver standard def as it's still the requirement up here in Canada with our MTV equivalent station Much Music) without ever actually seeing what their show looks like on a client monitor (i.e. an NTSC or HD televsion) until they take their locked picture to the online suite and only then do they get to see it on a TV...in a room that costs $500-$1000 an hour (i.e. they do this cuz like me, they don't have a high end video card or HD televsion). I just don't think it's wise to not see what your video actually looks like on an NTSC monitor (for all the obvious and not so obvious reasons I'm too lazy to list and you already know), hence why I downconvert my HD dailies at the offline stage (instead of downconverting my locked/color corrected HD timeline at the online stage) so I can actually see what my video looks like on an NTSC monitor (i.e. the medium it will broadcasted in) as I edit it.
- Because I shot some DV footage with the HVX-200 (emergency shooting only remember) I know needed to work with DVCPRO 50 23.98 and DV 30p in my FCP project. After coming to the conclusion I had no way of getting my DVCPRO 50 23.98 (downconverted from my DVCPRO HD 24pN remember) into a 29.97 timline, unless I laid it to tape (popular belief at the time), I unfortunately chose to upconvert my DV 30p to DVCPRO 50 30p and then reverse telecine it (pulling out frames from orignal 30fps source material, i.e. resulting in a slightly choppy look when played back) in order to work entirely in a DVCPRO 50 23.98 timeline. Another editor agreed that unless I wanted to lay my 23.98 footage to tape (the HD quicktimes or the SD downcoverted quicktimes) I should just make my 30p material 23.98 by reverse telecining in either cinema tools or a program he suggested called DVFilm Maker. I checked out DVFilm Maker, looked pretty mickey mouse to be honest and just reverse telecined in cinema tools.
- So, Mr. Shane, after I read your post that you were looking into what compressor can do, I thought it'd be worth a double check if DVFilm Maker could turn 23.98 to 29.97 and LOOOONG BEHOLD (wipe my eyes) it has a catagory called "Post-Edit Processing" that offers the feature "Convert 24P/23.976P to 3:2 NTSC". I quickly double checked my HVX-200 manual and sure enough, 24pNative mode records the "active a.k.a. effective" frames in the 3:2 pulldown process giving you true 24fps, so surely it could be REVERSE-reverse-telecined (also referred to as "pulldown" I guess). And it bloody well can. No recompression, no visible damage what so ever, just adds 3:2 pulldown and creates a duplicate 29.97 quicktime.
All my footage (whether it was HD 23.98 or DV 30p) is now DVCPRO 50 29.97 and can be edited without loss of quality (other than it being SD which was intended for now).
And it just occured to me, if you wanna save hardrive/P2 card space while in the field, why not shoot 24pN and then pull it down in post? (that was probably a loaded question that will generate a lot of responses entitled "Actually, you idiot..." but whatev. I'm so relieved at the moment that NO sacrifices have to be made. But I'll be damned if I didn't need 5x times the hard drive space than I should of for this little music video (i.e. all the original dailies in their original codecs HD/SD, and the duplicate sets of upconverted/downconverted dailies).
Goooodnight.

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 help

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