Sample Rate Convert Loses Timestamp

Can anyone confirm....
Sample rate convert a Broadcast wave file (with time stamp) from 44 to 48, The resultant 48k file has no time stamp info????
Ive tried this with copy/convert, bounces. and SRC from the sample edit window.
Anyone?

In reply to my own post.... Applemagic are you listening....
I have had it confirmed that when converting (in Logic Pro) a Broadcast Wave file from 44.1k to 48k the Original Timestamp info is lost
Moore

Similar Messages

Imported Audio Not Sample Rate Converted?

I received some audio files for a project that were accidentally sent at 48k instead of the 44.1 I need.
Isn't Logic supposed to automatically do the sample rate conversion if you have the little settings box ticked for "convert sample rates for imported audio"?
I can't get it to work. I use the audio bin "add audio files", and import the 48k files. They get imported, but remain at 48k, unchanged, after import.
Is this another bug or is there some part of the process I'm missing?

Import Audio File (Shift Command I).

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.

Multiple sample rate question

I have one project with edits of different files all over the place recorded at 44.1 kHz. I have another project with edits all over the place recorded at 88.2 kHz. I wish to combine all tracks from these two projects. I realize I will need to convert the sample rate of all the 88.2 kHz files, but I don't want to mix down any track. I want all of the edits to be cut at the same places in time, just with the referenced files being at a different sample rate.
I've tried everything. It seems that it is impossible. If I convert the sample rate of the referenced files, then open the project and change it's sample rate, the edits are now cut at differnt locations in time. It seems I have to mix down each 88.2 track, then sample rate convert these to 44.1.
But I need to have all the edited segments as they appear in both projects in the same project. Any ideas?

Don't record at 88.2! Why would you do that?!
I'd back up and archive your current session first as you are probably going to have to do some very destructive messing.
I haven't put this to the test but maybe you could try converting the route files in a third party software like ProTools. You'll get a bunch of prompts when you reopen Logic but this may work. But for me the simplest (albeit time consuming route) would be to convert all of your regions to new original sound files (you can do this from the audio pop down on the arrange page). Then in your audio pool select all unused audio, delete and go through each new file and convert the sample rate. It's a bit of a car crash situation needless to say next time preparation preparation preparation!
Finally when you have been through this kind of process make sure that you are operating at the correct new clock rate before continuing otherwise this stuff still aint gonna work.
Hope this is of help.
Good luck.

Audio sample rate 48048 incorrect time code

When importing Production Audio that is recorded at 30NDFTC and 48048 the time code displayed in the bin is incorrect. It looks to be giving an extra pull up to the time code. For example the wav file header time code stamp is 10:17:56:26 the time code in the bin is 10:17:19:27. Seen this in all versions of FCP 5.1.4 through 6.0.2.

Yes, this is a feature film shoot, The DEVA 5 can, of course, record 30 fps NDFTC and you can set the sample rate to 48048 instead of 48k with a pullup. With the recording set to 48048 Final Cut wants to add an extra pull up. With the recording set to 48k Final Cut acts just fine.
Yes, the way this method works is that Prod Snd records at 30fpsNDFTC/48048H in the field/studio then in Telecine when it's pulled down to match the 24fps film that is now 23.976 the audio will then be transfered at 29.97fps NDFTC/48kH. This avoids need for a sample rate converter that would be needed if the Telecine is going to an HDCAM or SRW for example. A Fostex DV824, for example, can easily do this in Telecine.
I would like Final Cut Pro to do the same. So that if I want to just Telecine the film then post sync picture and sound in FCP it will line up correctly.

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.

Converting 24-44 sample rate to 24-48 - without losing video sync?

Hi,
I am doing a video shoot for a shot film and I need my final outputs at 24bit and 48khz for audio. Video frame rate is 24fps.
I am planning to do live sound recorded externally for good quality when compared to camera internal records. But I have a macbook which is 9 yrs old and I can only use garageband with my focusrite saffire6 usb audio interface and I cannot use logic in this old macbook.
So my questions are
1. Can I record live sound using garageband as DAW and export files at 24-48 from GB? I find it can do only 24-44....
If not...
2. Can I open the garageband project in my imac at my studio with (Logic Pro X) later at the end of each day of shoot , and export the recorded tracks of GB to 24-48 wav or aif from logic pro x ? ( I dont want to change the logic pro x project to 24-48 from 24-44 as the pitch of the recorded voice alters.
If i export the tracks at 24-48 from a 24-44 project and then re-import it to another logic pro x project (24-48) what exactly changes ? Will i lose video sync / lip sync ? will there be any noticable quality loss ?
Pl. do help me with my above questions. I will be converting small small audio clips of the shots related to the video.
I dont want finish my movie then change the sample rate of the entire audio track....
So will the video / lip sync be lost in any way ?? Is 24-44 to 24-48 convertion recommended ?? as I have only my old mac book only to use to record audio on the shoot location and i dont want to carry my imac to the location... I will be using my rode video mic pro and saffire 6 usb interface in my old macbook i am not in a position to invest on another recorder.
Looking forward for your replies and comments. Thank you.
Kind Regards,
Arun Kanth

Hi
Nik on Logic wrote:
I know 48 is the norm ..they say for video.. but i am working only with audio here that I am delivering.
You are delivering audio for a video project.
Nik on Logic wrote:
I am going to deliver a long bounce which then the director will put in his system and mix with the other elements of the film. dialogue, etc.. I was not asked for a 48 file .. and in the past I was always fine by delivering 44.1 audio files to editors or mixers..
Arguably, you should've asked what format they require the finished files to be in (ideally before you started). Probably best to check now what they can/will be able to work with. Then you will know what must be done with the work you have in progress.
CCT

Converting Units in LabView and specifying sampling rate with Universal Library functions

Hi,
I am having trouble converting units in LabView 7.0 and having it write to a file / output on a chart in Nm instead of in volts. I can't seem to find any straightforward instructions, eventhough it seems like a simple task.
Another task that would seem rather simple is changing the sampling rate with the XAIn program from the Universal Library for LabView (this is version 5.40).
If anyone could help me out, I would greatly appreciate it - I have tried other sources without much luck!
Thanks
Jenna

Are you really needing to change units or do you need to scale the voltages read from your sensors? If it's scaling, it would be trivial to set this up in mAX or with the DAQ Assistant if you were using an NI board. Since you are not, then you could use the Scaling and Mapping Express VI. The y axis of a chart is just a label. You can use the text tool to change it to anything you want.
I have no idea how to change sample rate for your board. Have you asked the vendor?

Converting sample rate when exporting

CS5.5
I'm using Audition to do some final mastering to 16-bit stereo 44100 wav files, in 32 float multi-track sessions.
All I'm doing is increasing the clip gain, then exporting to 16-bit stereo 44100 wave and 320 mp3.
I'm not changing the sample rate, channels or bit depth, but I am applying triangular dither when exporting.
The thing I've noticed is even though I'm not changing any of the properties of the files, when exporting Audition still tells me its "converting sample rate", despite the sample rate being the same as the source (44100). The quality advanced tab on the sample rate is greyed out.
Also, when I check the history box for the newly exported files it tells me "convert sample type on save as"
What is Audition doing at this stage?
How can it be converting the sample rate, when the sample rate is the exact same as the source? And the quality tab for sample rate conversion is greyed out?
All I'm doing is upping the clip gain slightly and then applying dither to these files. No sample rates are changed.
Any help greatly appreciated.

brazil101 wrote:
I'm using Audition to do some final mastering to 16-bit stereo 44100 wav files, in 32 float multi-track sessions.
I'm not changing the sample rate, channels or bit depth,
Except that you are...
If you open a 16-bit file as a 32-bit float, then you've changed the bit depth. It's the right thing to do if you are doing amplitude changes, as it makes any change effectively lossless, but it will still have to be converted back to 16-bit if you elect to store it like that.

Burn voice mp3 play list to audio CD. convert bit rate or sample rate.

I have several audio files (voice), 30 in all, that I wish to play in a car CD player.
The car player cannot play mp3. Using iTunes I tried buring a play list of mp3 files to an Audio CD.
I get an alert from iTunes that I may require multiple CD's. Each mp3 file is around 90megs in size or about an hours worth of audio (voice).
The audio files are voice no music. But they are at 192 kbps bit rate and 44 khz sample rate.
Is there a way of converting them to squeeze more files onto a CD? Changing bit rate or sample rate?
Can itunes do that? Or is there a utility that can do that?
thanks

Now that memory serves me right. CDs have the MB size and the Minute size on them. example 700MB/80Minutes.
How we forget...
Thanks though

Converting sample rate enmas

I have built a rather long and elaborate document in the style of Old-Time Radio.
It consists of narration at 48 sample rate,
and 200 sound effects from Apple's content of which about 95% are at 48 but some, especial the music and at 44.1.
Is there any way to, and what is the easiest way to convert almost 200 regions (or their underlying audio files) to ALL 44.1?
(BTW, in the document-preferences, it is set to 48, (it must have defaulted that way when I first started, sorry I didn't change it from the start.)
If I have to convert these almost 200 files individually, I won't do it, but ideally I wish I could convert these to 44.1.
Or should I just leave it at the superior 48 until export time?

Hello,
It would seem that I'm stuck in 48 at least for this document.
Here is what I've discovered if I change the document sample rate from 48 to 44.1, and then play back the audio files already in 44.1...
SOME of the files sound effected.. but others sound just the same.. BAFFLING.
also if I change the document sample rate from 48 to 44.1, some of the regions get moved!!
It creates at mess.
Will stay with 48 on this one.
Sure wish I know why only some of th 44.1 files were effected.

Convert Sample Rate on Import- does not work

Setting: *"Automatically convert sample rate of imported audio"*
You'll want to set this if you have audio on your computer
that wasn't sampled at 44k. It prevents your imported audio from
sounding like Mickey Mouse drinking a gallon of espresso and inhaling helium.
HOWEVER...It does NOT work when you drag audio into the project from
Logic's internal File Browser.
Dragging audio into the project from Logic's Browser,
or browsing to the file in Logic's Browser and choosing to
"Add selected Files to Audio Bin",
will NOT convert to the audio's sample rate.
Sample rate of your audio is only converted if you do: "Import audio file",
then browse to the file(again) in a popup browser dialog, and then choose import.
If the audio is already in your Audio Bin, choosing "Convert File"
will bring up another Finder Dialog asking you where you want the
converted audio to be placed.
Now you'll have to Browse (yet again) to the song's folder, because that's where
you'll probably want it anyway, and choose to put
your converted audio into the song's folder.
Please realize: If a composer or producer wants his/her audio to sound like
Mickey Mouse on Helium and Espresso, he will do it after the fact, on purpose, using a plugin.
Is it too much to ask, to have this Setting do what it says it does, with ALL audio brought into the song?
No matter how I bring audio material into a Project, the sample rate should always conform
to the Project's sample rate.

Hi there
I've been using Captivate since version 1. And to be honest, I'm not sure I've ever seen the option work. If it does work, I'd also like to know under what conditions it works. Perhaps it only works with certain applications. I'm not sure. Largely I've ignored the option exists because I've never recorded anything where I wanted tooltips to be an option. I've never wanted to simulate to that detail.
You should probably consider filing a bug report.
Cheers... Rick
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Q: Convert audio bit & sample rate during import

I have a project that's in 24bit/48KHz, and want to import an audio file (mp3) that's in 16bit/44.1KHz.
When I import the mp3 file into the project either through drag & drop or the Add Audio File dialog box, it doesn't convert the file into the project's native bit & sample rate. Consequently, when I play it back in the Arrange window it plays in chipmunk voice.
I thought Logic used to automatically convert audio files into native resolution. How do I do this in Logic 8?
Thanks,
Steven

Thanks for the response. My project currently does have those options checked, and it's still not converting.
To double check, the mp3 file does get copied to the project's Audio Files directory. It also creates a file with the same name and the file extension .ovw - but doesn't seem to be converted.
Any ideas? Thanks again -

Does itunes convert sample rates before burning to disc?

Question:
If I set up a playlist that incorporates tracks that have a 48kHz sample rate, does itunes convert to 44.1kHz during the process - or is the pre-existing sample rate retained?
Thanks.
-ptfigg.

I think it leaves it as-is if you burn as an mp3 or data disc. As an audio cd I believe it converts them to audio cd standards.

Sample Rate Convert Loses Timestamp

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