Mixed Sample rates in STP

Last night I was working on sweetening a video project and ran across a problem that certainly others must have encountered. I was wondering what others do in this situation.
I sent a FCP project to STP that had some library music in it. It was a 48K file with some 44.1 music in it. FCP resamples on the fly so the mixed rates was never a problem in FCP.
And when sent to STP, the program played back fine (not sure why) during the sweeten and the mixed sample rates was NOT a problem UNTIL I "sent" it back to FCP. Then the music was "pitched up". Just a bit faster and few seconds shorter (an obvios resample error)
I can send the mix back without the music and do the final mix in FCP, but I wonder if I'm missing something. Looking for a more elegant solution.
Thanks for the suggestions.
David

Forget about mixing rates in STP2 anymore. It's a no-brainer for a professional audio software, yet STP2 manages to completely bork it.

Similar Messages

I am finding FCPX has trouble with mixed audio sample rates

I am just putting this thread out there to ask if anyone has experienced issues with mixed sample rates in the timeline? Specifically I have noticed that a timelilne with an audio rate of 48KHz with have troubles with a song at 44.1KHz - this is irrespective of what sort of video codec I am using. It happens to me a lot lot with stills actually.
If I preconvert to 48KHz, I do not have the same issue. Just asking if this has been discussed or mentioned elsewhere so I can figure out if it is just me.
Thanks

You might want to duplicate the capture preset for DV/NTSC 48k and change the audio to 32k.
Then try that.
But I'd also change the JVC's audio settings to record in 48 while you're waiting. Should be able to do it... cheap or not.
CaptM

Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000. Is this possible in Audition ? I have been going through all the preferences and all the tabs and I can't find this option. All I find is a convert option, or the adjust option. But that's not what I want. I want to mix down this way.
The closest thing I found is when I go to "Export Audio Mix Down", I can select 32 bit. Then there is a box for sample rate, with all the different values. But it won't allow me to change it from 44100.

JimMcMahon85 wrote:
Can someone explain this process in laymens terms:
http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
Second point: you cannot dither a 32-bit Floating Point recording, under any circumstances at all. You can only dither a recording if it's stored in an integer-based format - like the 16-bit files that go on a CD. Technically then, you can dither a 24-bit recording - although there wouldn't be any point, because the dither would be at a level which was impossible to reproduce on real-world electronics - which would promptly swamp the entire effect you were listening for with its own noise anyway. Bottom line - the only signals you need to dither are the 16-bit ones on the file you use for creating CD copies. And you should only dither once - hence the seemingly strange instructions in the Ozone guide about turning off the Audition dither when you save the converted copy. The basic idea is that you apply the dither to the 32-bit file during the truncation process - and that's dithered the file (albeit 'virtually') just the once. Now if you do the final file conversion in Audition, you need to make sure the dithering is turned off during the process, otherwise all the good work that Ozone did is undone. What you need to do is to transfer the Ozone dithering at the 16-bit level directly to a 16-bit file proper, without anything else interfering with it at all. So what you do with Ozone is to do the dithering to your master file, and save that as something else - don't leave the master file like that at all. After saving it, undo the changes to the original, in fact - otherwise it's effectively not a file you could use to generate a master with a greater-than-16-bit depth from any more, because it will all have been truncated. Small point, but easy to overlook!
just what's the easiest way to test if a simple dithering setting is
working for 32-bit down to 16-bit in Audition?... Why is there no info
about dithering from 32 bit to 16 bit (which is better then dithering
from 24-bit isn't it)?
I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
It's all about educating people in the end - and as you are in the process of discovering, all education causes brain damage - otherwise it hasn't worked.

OMF - Mixing Different Sample Rates

Hello -
So, I've got an OMF file from a FCP project that I opened in Logic and have been mixing without any issues. Interestingly, all of the audio files associated with this project are 44.1 kHz, but I am mixing in 48k. But everything is right (sounds right, looks right, syncs with video correctly).
But... if I open the same files in an external editor or quicktime, they play back incorrectly. And if I save a file in a different program, even if it is still at 44.1, and bring it back into logic, it plays back incorrectly. This is problematic if I need to edit an audio file somewhere other than within logic (say I want to do some noise reduction in soundtrack pro).
Anyone run into this issue or have any ideas about how this happened?

There are a few ways to look at this.
1) Regions in the arrange all play back at the session sample rate. Example: 44k session, 96k audio file in arrange=slower playback
2) Logic automatically converts output sample rate so you can record independent of CA devices. Example: You have a session recorded at 96k, your interface is not connected, Logic will load Built in Audio, Logic runs tyhe sessions at 96k and converts the SRate of the session to match the supported sample rate. So you can run sessions at unsupported sample rates, this rarely makes sense if you cannot capture your audio at session sample rate (if needed).
3) There are a few other options for handling this, such as EXS24, which automatically handles SRC.
4) When Importing Audio Files there is a song preference which you can en/disable to automatically convert SR upon import.
I think your friend may have referred to point #2 and it was interpreted as point #1...perhaps. Hope this clears things up. J

Mixing audio sample rates

I'm making a sport video, and I'm using DV tapes from another parent. He has a different audio rate setting on his tapes. When I capture them, I get this warning message about sample rates setting. When I view the tape it seems fine.
Is there a work around this issue when using other people's tapes?
Hank Kearns

Hello Hank...
When capturing media, it's important that your capture settings match the origonal media parameters ie- if the footage/audio is at 32Khz, for example, then you should capture at 32Khz... otherwise you run the risk of video/audio sync loss.
If you have 1 tape which was recorded at 32Khz, then another at 48Khz, you should change the capture settings for each tape accordingly.
Once the media has been captured and is in your browser, you set up the timeline to the setting you desire.
So... you can have a browser bin with media of both 32 and 48Khz.... then a sequence/timeline of (for example)- 48Khz. When you place the 32Khz media on a 48Khz timeline you just have to do an audio render (sequence>render all/render only> ITEM LEVEL), then you will be ok.
You say that the footage seems fine, but the more you work with it, you will probably experience sync issues... you might not notice them at the top of the timeline, but further down, things will slip.
So... you can mix clips with different audio rates on one timeline, and just audio render to get it stable. The MOST IMPORTANT thing to do is ensure your audio rate capture settings match the audio rate on the origonal tape.
Make sense?
Good luck
J

44.1 versus 48KHz sample rates

Since manufactured CDs are 16-bit, 44.1 KHz, is it better to record tracks in that format? It looks like STP defaults to 24-bit. Is it better to use the higher sample rate as well? 48KHz?, or should it be even higher than that?

Hi Bob. Let me see if I can remember what I learned way back in my college physics class. Keep in mind there are almost as many ill-informed opinions as there are people who record digital audio. (Oh! Dig!)
Some of that stuff sounds silly to me. Generally it's wise to use the best quality that's available to you unless you have space or bandwidth constraints. Saying you should do all your recording at 16/44.1 because it's ultimately going to end up on a CD is like saying you should shoot all your digital camera pictures at 320x200 because it's ultimately going to end up on a web page. Shooting pictures at higher resolution allows you more flexibility when editing them down for the web site, right? And audio is no different.
Here's some basics, explained in pictures:
http://www.musiciansfriend.com/document?doc_id=88273&g=home&src=3SOSWXXA
You can find more stuff like that (and endless hand waving and opinons) if you google "Bit Depth Sample Rate Physics".
Bit Depth:
Imagine in some simple world your recording software could sample waves that had amplitudes between -1 and 1. With 16 bit samples, you can record 65,536 discrete levels. (You might define a sample value of 0 to be -1 and 65,535 to be 1.) At 24 bit, you can record 16,777,216 discrete levels. The resulting representation of the wave you record will be far more accurate.
Sample Rate:
This comes down to Nyquist Frequency. The idea is that you can only record frequencies up to exactly half of your sample rate. That doesn't mean you can get a good or accurate recording of those highest frequencies, though. The higher your sample rate, the more accurate picture you can get of those higher frequencies.
This is important for recording and playback, but it's even MORE important when you later go to combine this recording with other recordings. Mixing signals, the mathematical results of processing with effects, etc. All those things will give you better results if you give them higher resolution going in.
Combining 16 low resolution tracks will give you a much worse result that combining 16 high resolution tracks and then down-converting the result. The combination of the 16 high resolution tracks will be a much more accurate representation, right?
That said, there are certainly diminishing returns. 24 bit/96kHz can give you great results, but will take up more disk space and processing bandwidth than 24/48. If you're not using superb mics and preamps, the improvement might not justify the difference. You might consider trying some experiments to see if you can detect differences yourself.

Sample rate off the audio input and out devices do not match - what to do?

This is fundamental, I know, but nevertheless I can't find my way around it. I get this error message when trying to recor:
Sample rate off the audio input and out devices do not match
and am asked to do this:
Use the appropriate operating system or audio device control panel to adjust the sample rates of the input and output devices to use the same settingt.
I have defined the sample rate to 44.1/16 bit in accordance with my inbuild soundcardt.
I am trying to record from LineIn.
When running on a M-Audio sound card I don't face any problems.
HP 8560W, sound card IDT/High definition audio Codec
Any suggestions?
Knud
Copenhagen

You're sure you have set BOTH the input and output settings to 44.1 16 bit?
Which version of Windows are you running? There are a number of posts on this forum about how to fully access both the Windows Mixer and the Mixer for your soundcard. Especially, you need to ensure that all "Windows Sounds" are turned OFF.

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.

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!

DIgital audio in sample rates, 441 48 96 but no 882... is there a way to ad

When feeding my macbookpro a digital audio signal via optical cable while viewing the audio midi setup utility, it shows that the mbp's internal clock automatically switches to the incoming sample rate. I got it to work with 441,48 and 96k. When I fead it the 882 sample rate it automatically switches back to the mbp's internal clock and reverts to line input instead of digital. So the audio midi setup utility shows when it is in digital input mode the three sample rates 441,48 and 96k in the format pop up window and not 882 so it must be just a software issue and I am wondering if there is a way to add the sample rate to the list. If it can take in 96k it should also be ale to take in 882. So does anyone know how you could ad a sample rate of 882 to the list in the audio midi utility format list?

One added note. The reason that I want to be able to do this is I like to record my audio projects at 882 and if the mac book pro could except the sample rate I could then use it as a 882 digital 2 track recorder. Enabling me to skip hitting the bounce to disk function in pro tools and actually be able to ride a fader as the mix is printing to the macbookpro, where as when you bounce to disk in pro tools you cannot access the mix as it bounces to disk.

Problems with sample rate

hello:
I have a sound blaster audigy platinum ex and I'm trying to make some recordings with adobe audition, but this program works at 44khz and the sound card seems to work only at 48khz, so it doesn't let me recording. how can I change the sampleing rate of the card?
thanks.

By setting it in Audition audio settings. W/ ASIO device drivers you can have 16-bit/48kHz (and IIRC, 24-bit/96kHz) mode(s) only). For other modes you need to use other device driver mode. Also, Asio4All should handle all modes (16/24-bit and 44.1, 48 and 96kHz).
- http://www.asio4all.com
The quality in 44.1kHz mode is POOR for your card. If you switch to 24-bit/44.1kHz quality is otherwise better but the frequency gets cutted already around 15kHz.
You get the best quality by setting the sample rate to 96kHz (16 and 24 bit resolution are ~same in these cards) for recordings (for project?) and then when export the final mix just use SRC to CD quality (Audition do have quite good SRC quality but, if you use additional software for SRC (like Voxengo r8brain (free)), export the final mix using 96kHz and then execute SRC).
Something more on Creative ASIO -
http://forums.creative.com/creativel...essage.id=1726
jutapa
Message Edited by jutapa on 11-09-200602:43 PM

Transferring music recorded at 32kHz sampling rate

In my project transferring music from DAT tapes to HD, I've run into a problem. A few tapes were recorded at 32kHz sampling rate for longer recording time.
STP can record at 32kHz so in theory I can transfer music digitally. But there is no sound when I monitor the tape and the Audio MIDI setup doesn't have a 32kHz option to choose for recording.
Can someone steer me in the right direction to get this music off the tapes? Is recording to the line analogue input the only way?

The problem is not with STP, the problem is with the hardware. if you want to record at 32Khz, you'll need an audio interface that is 32Khz compatible and I don't think the Mac's built-in audio is. Otherwise, we could select that in the AudioMidi Setup.

Why Audition CC don't show sample rate correctly?

Hi
Using Audition CC on Mac OS X with Mackie 1220, 12-channel premium analog mixer with integrated 24-bit/96kHz Onyx FireWire I/O (Quality is amazing!)
Need to record on mono a horrible old tape recording with voice only to "hero"'s repairing audio and configure the best sample rate for recording.
My preferences Audio Hardware on Audition are:
Device Class: CoreAudio
Default Input: Onyx Firewire (0025)
Default Output: Onyx Firewire (0025)
Clock Source: Device
I/O Buffer size 512 Samples
Sample Rate: 96000 Hz
Question 1: Expert said do not use more than 48000 Hz for recording voice only and use mono. Humans can't listen more than 20000 Hz... well... My recording on 96000 Hz is better than 48000 Hz and I'm not Superman... and can hear better difference with 96000 Hz recording. Don't know why.
Question 2: Why after selecting Audio Hardware preferences to 96000 Hz after recording the File Panel said my track is 48000 Hz Mono 32 (Float)? My hardware can record at 96000 Hz and preferences is set correcty at that Bit Depth then WHY don't appears 96000Hz on File Panel after recording and instead appears 48000Hz?
Question 3: Don't understand the differences of my 24-bit/96kHz Onyx FireWire I/O and 32 bit (Float) of Audition. If I record using 24 Bits (Maximum bit depth of my hardware) why appears 32 bit (Float)? My recording are 24 Bits or 32 Bits?
Thanks :-)
Tom

CS6 has a serious issue with saving files correctly. The program is asuming that 48kHz is the maximum you will be using and in my case it saved a 96kHz recording with a 48kHz internal header. The file size is consistent with all my previous 24/96 recordings and it sounds just fine interpreted correctly - but played an octave low in frequency and tempo it really sucks unless you are a Blue Whale..
I can play it "interpreted" as a 192kHz file just fine, and it now sounds 100% right, but I cannot save it correctly. I have yet to find out how to recover because when I use "convert Sample Type" it saves it with the same mistake - the wrong sample rate off by the same ratio again.
It is a program flaw - so at this point you cannot record with sample rates higher than 48kHz in CS6 and depend on your file being OK.
Tom the reason you can tell the difference between sample rates is that your hearing has two dimensions, frequency and timing, I sure hope you can't hear the bats at night but I expect that you can tell a good drummer from a bad drummer. In addition there is also the issue (dimension) of bit-depth - instrument decay and acoustic space occupies the time space between notes and if it is not sampled at the right time and place or rounded off to the nearest digital digit you have a problem.
You all know that some humans have perfect pitch and others dont, this gives you some indication how much we each differ. Some people have even learned to use echolocation; the best know cases being blind people because they are not supposed to be able to find their way and know where they are. You can learn underlying concepts of this little discussed aspect of human hearing here and hit the university libraries for the rest. http://en.wikipedia.org/wiki/Human_echolocationhttp://
SteveG's comment are only true in one of the three dimensions - frequency range. We all hear about the same for starters anyway.
The second dimension, timing, is what provides spatial information. Coarse sampling effects this as well. My most recent experience was when transcribing some old casette tapes - when I experimented with FLAC and WMA lossless I found that they din NOT downsample well. Spatial information was significantly deminished be that aucoustic or studio work. Needless to say this surprised me because I was auming that I could drop the samplerate to 48kHz as soon I was done editing and save a lot of drive space. For now I made some excellent 48kHz/24 bit mp3s (320) because they altered the sound the least.
Now you know some additional reasons why older guys like me who have lost the high end and a lot of decibel as well can still tell the difference:we have good sense of timing.
Anyway - I need to learn how to edit my files to reset the sample frequency header - real fast. I just recorder a fabulous Madrigal group for 2 hours and my files are lost in the vortex.

On the fence for which sample rate to record at (44.1 vs 96)

Been reading tons of posts on the sampe rate debate. My friends (across the country) and I are about to start to collaborate on the great American rock album that we didn't quite get right back in the day in college. I'll be running the show sending them scratch tracks with clicks so they can lay down individual tracks and I'll import them.
I'm torn on which sample rate(s) to use -- and want the best quality possible, of course. I've boiled it down to the following pros per sample rate. Any advice/comments much appreciated. thanks
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
- Smaller file size to post/share (Dropbox cost per/storage size)
- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
thanks!

rcook349 wrote:
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
- Smaller file size to post/share (Dropbox cost per/storage size)
- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
thanks!
44.1 kHz still is pretty much standard for MP3's.
Your friends/collaborators can pretty much use any application that can record PCM (or even MP3) audio; even if they're not playing to a steady tempo, you can line everything up in Logic, with flex.
Using Garageband and one set tempo should also work. Just remember that you cannot open Logic files in Garageband, only Garageband files in Logic. The Audio Files recorded by either, can be used (imported) by either.
Higher sampling rates will not "future proof" anything. In fact, that whole concept is flawed. Your best bet for now is simply 44.1 kHz 24 bit uncompressed PCM files in their most widely used form: AIFF or WAV.
96 does not noticeably sound better than 44.1, unless you have a top end interface and a very delicate and very complicated mix, and admirably acute hearing. In some interfaces 96 or 88.2 have been found to sound worse than 44.1, because of clocking inaccuracies getting progressively worse at higher sampling frequencies. I would stick to 44.1, it has lots of practical advantages (as you pointed out), and the sonic difference with 96 kHz is marginal at best, and certainly not worth the price: "double" rates need double the CPU power for any plugin processing. That's the biggest loss. Half a Mac.
Bitdepth on the other hand does make a significant difference. There is no reason not to record everything at 24 bits. Shorter: always record at 24 bits.
O, also just spotted your remark about Logic not "letting you" bounce MP3/M4a to 44.1 kHz. You must remember incorrectly, because I never bounce MP3 or AAC to any other frequency than 44.1 kHz. However, it may be that this rate is tied to the projects' sampling frequency as set in the project settings, and the last time I used 48 kHz was in LP 8. I'll check that now.

Audible glitching - audio sample rate mismatch?

I'm capturing some old DV tapes for a friend who no longer has a DV camera to capture from. I've captured into FCE4 and the video looks fine, but all the audio sounds like a sample rate mismatch, that static-y glitch sound that I've heard before in audio programs when there's a sample rate problem. The audio on the captured clip is 16-bit, 48KHz, big endian.
I'm using a sequence with the DV-NTSC preset with 48KHz audio. I made sure to set it up in Easy Setup beforehand so it would be the same as the sequence. The camcorder I'm using for capture is my old Sony TRV-70 (have also tried my TRV-50). I can't get either camera to tell me what sample rate the tape was shot in; all i can get it to tell me is 16 bit and SP mode. I'm assuming this will be 48KHz?
When i try capturing with the DV 32KHz easy setup (and sequence) it gives me the error about the sequence rate not matching what's on the tape, and it sounds worse.
I don't know what camera originally shot the tapes. But the audio sounds perfectly normal coming out of the speaker in my sony camcorder.
Am I missing something?

16 bit and SP mode. I'm assuming this will be 48KHz?
That's correct.
Make sure the Sequence you select is a 48 kHz choice abd double check that you do not have a PAL/NTSC mix up.
Al

Mixed Sample rates in STP

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