LogicX - Maverick - Sample Rate issue with Audio Interface

As a test prior to updating my Mac Pro and iMac, I loaded ML so I could test out LogicX. I've been running Logic 9 either using the built in audio or a Line6 guitar Port. The MBP is a late 2009, non retina, 13 in with 4 GB Ram. As compared to my iMac, the laptop can barely play back my tracks which are mostly software instruments (I don't record audio until the midi arrangements are finalized).
So, I updated to ML to try out X and of course, had to deal with the 32 bit plugin issue. Luckily, 32 Lives and Vienna helped out with 90% of my plugins.
Now, I updated to MAV and had to go through an arduous process of re validating the AU plugins for both Logic 9 and X.
After doing so, I loaded up a song that did have a single bass audio track. I updated the Guitar Port driver from Line 6 to the latest driver. The song will start, but then I get a stuttering and an error message about running the audio at a weird sample rate, eg 28654. I tried using the built in audio and same issue.
I opened the song in Logic 9 and it works.
Any suggestions? Do you think one of the resurrected plugins is causing this problem? Everything worked OK in ML though.
Thanks and appreciate any help.
Todd
trademarkauthority

Let me start over ... the typical audio setting that FCP works with is 48khz, 16 bit stereo. If you check the sequence settings, it will probably show that to be the case unless you've created a sequence with other settings. The audio of any clip you bring into FCP really needs to match the sequence settings. To check the sequence settings go to Sequence->Settings.
So far, you've confirmed the clip is 48khz sampling rate but you didn't say what it's bit depth was. In FCP, right click on the clip and select "Item Properties." That will tell you the information I've been asking about.
-DH

Similar Messages

Project sample rate vs the audio interface clock

I used Audition for digitizing archival audio and I mostly work in 96 kHz because it's the current archival standard for analog audio, however I occasionally need to transfer DAT or ADAT tapes at 44.1 or 48 kHz. One thing I noticed when I went to change the project format was that my audio interface clock did not change and match the multitrack settings. I was able to change the clock manually but I had to go to the Audio Hardware preferences.
I find this behavior for a DAW very strange. If I have a multitrack open of one bit rate and my clock set to another, how does that work? How does Audition 6 play and record audio at differnt sample rates than the interface clock? When I open a multitrack that's 96 kHz and my clock is set to 48, what sample rate am I getting? The audio file being saved says 96 kHz but is it?
Audition 6
RME Fireface UCX
Mac Pro mid-2010

loneraver1 wrote:
If true, this is indeed strange. I have used just about every DAW under the sun for the last 10 years based on what my employer purchased and I have never seen a DAW allow you to record sample rates where the project didn't match the incoming clock.
There should be an option that locks the audio interface internal clock to the opened project. It makes it annoying any time I have to switch between recording in 96 kHz to 48 kHz. That's annoying on it's own right, but what's really annoying is that it's an extra step that I have to train our archival technicians who don't have a techical background in audio.
I can definitely say it's true for my set-up!
Surely, the "extra step" you refer to is not necessary if AA is resampling the audio to match the desired sample rate?
As for the "option" to "lock the audio interface clock to the opened project", for those cards, like mine, which cannot have their sample rate changed by the audio software, how is that going to work? I suspect this was how AA CS5.5 and earlier worked: unless session/project and interface sample rate matched, no audio. I much prefer the current "resampling" method; no necessity to have to think about soundcard/interface settings every time a different sample rate is being used.
FWIW, if I want to record something at 44.1 I always ensure the card and AA settings are 44.1; but for "playback" (or video editing which I do quite a lot of and which requires 48 - my video editor software will not "play" unless the audio sampling rate of the card is 48 - ) 48 is my "default" setting.
JMO!

MiniDV audio sample rate issue

I cannot get the audio sample rate to match. I have captured miniDV footage (32mhz/12bit). In Sequence settings the audio options for bit are 8, 16, 24. My audio is out of sync!!!! What can I do? I have 2 hours of raw footage and can't proceed! Any help is VERY appreciated. thanks in advance!

Hi again!
Hum ... that´s not as simple as it could be!
That way i´ll have to capture twice?!
There must be another way of doing it , i don´t think Avid can do it and FCP don´t .
As i told you i use both systems and i know they´re limitions (or i think i know!), but it´s strange because Avid Xpress DV 3.5 (from the age of stone ) do it in a blink ... ok ... found an FCP limitation!
Thank you!
If there´s another way ... please feel free to post it!
(and is was supposed to "reply" a "miniDV audio sample rate issue.
HBars

External sound card sample rate issue

After installing Yosemite I am having problem with Sample Rate settings on my external audio interface. It was working perfectly before the upgrade.
The issue is that the sample rate on my system is set to 44.1 Khz. When a new application attempts to access the audio card it plays but breaks up. This is indicative of a sample rate issue so I tried to set the rate to 48 Khz and then back to 44.1 Khz. Once I do this the system works properly until another application tried to use the card and I have to repeat the same process over again. While it DOES work it is not correct functionality and there is obviously something wrong.
At first I thought is was just my system until my friend also commented on it happening to him too.
I am using an RME Fireface UFX with the latest drivers issued on October 8th of this year.
My friend is using an M-Audio unit and he is finding the same issue.
I have already checked everything in Audio / Midi Set-up control and reinstalled the drivers for my audio interface. I also looked for any new settings to Yosemite to see if there was something new that might affect this. No luck and no leads on where to look for the next option.
Anyone have any ideas?

Hi,
The sound card do not support all sample rates. Which sample rates your sound card support you will find in the the manual.goto help in the toolbar select find examples and search for sound Open the Sound Input to File.vi file.This will give you a template for recording sound.You have to set the sample rate then open a sound file for write.
Please post a sample VI that you have created.
also,please go through this:-
http://www.zeitnitz.de/Christian/waveio
Thanks as kudos only

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.

My firewire port works with ext hd but not with audio interface

2010 Imac 21.5 running 10.8.5. My firewire port works with ext hd but not with audio interface. The local authorized Apple tech said that I need a new motherboard, but my firewire ext drive works perfectly. I don't understand. Thanks.

You must go to the website of the audio interface manufacturer and look for, download and install updated drivers for the audio device you are using.
This is not, neccessarily, an Apple issue.

Experiencing issues with audio scrubbing in Premiere cc 2014

Since upgrading I've been having issues with audio scrubbing in Premiere CC 2014.
Some of the time it won't scrub at all (silence) when I know that scrubbing option is turned on. Other times it will scrub the wrong section of the timeline, I'll hear dialogue from a different point in a clip.
Much of the time I experience this I'm working with multicam clips in sequences.
Working on iMac 2013, 32GB RAM, NVIDIA 780
thank you.

Hi IP,
Intensity pro wrote:
...Adobe are you listening?
Yes, but it's Saturday. And I'm on vacation. You know??
Sorry that we do not have 24-7 support. The best time to catch us is 7AM-7PM PST, Monday to Friday.
Intensity pro wrote:
Currently using / testing a 30 day trial of CC 2014 (CS5 user for years with no issues) and all was fine yesterday when editing. Turned on today and had no way to scrub audio. Also had no Intensity Pro external monitor which worked fine the day before. Restarted PC and still nothing so starting poking around with the playback settings in preferences by choosing Adobe Desktop audio in the audio device section choosing OK then selecting back to Blackmagic audio then hit Ok. All of a sudden there was my external monitor and was able to scrub audio again. It's like it's sleeping knowing it should work until you poke it a few times to wake it up. How my Intensity Pro external monitor and scrubbing audio being related in any way is perplexing. Or am I being obtuse?
I would contact Blackmagic Design to make sure your drivers are current. Can you try that?
Thanks,
Kevin

Issues with Audio

Hello, I'm Having an Issue with my Audio in AE. When I try to Shift-RAM preview my footage It doesn't play any of the audio from the Footage. The video itself plays fine, but no audio. When I ad it to render que and publish it, it plays audio.
For the record, the footage is not muted, and if I put in just an audio clip it plays it fine.
Thanks!
Jason

Have you really checked that the audio is not muted under Shift+Ram Preview Options? It has to look like the right image.
I'm just asking again because when it's muted and you scrub the timeline with control key pressed down you get audio even when it's muted in the Preview window !
and don't push the play button (plays without audio) push the RamPreview button |||> (the one on the right side)
If you did everything right and still have no audio then take a look into Preferences/Audio Hardware and change the audio driver or the sample size buffer to see if that helps.
The default 'Sample Size Buffer' value for the driver 'After Effects WDM Sound' is 2048.
If that don't help - I'm out of answers
Good luck

Sample rate/quicktime pro audio/video

Hi all:
Have a few questions related to Quicktime pro on mac related to sample rate of audio and video...any help understanding much appreciated...
I am a music composer using logic pro and recently started working with filmakers who send me quicktime files of the video to score to...
regarding sample rate of audio as played back in quicktime...I'm a bit confused....
When playing back either an MP3,wave,Aiff at 44.1k or 48k
seems that quicktime plays at the right speed and pitch of audio...Does Quicktime use it's own audio processor to convert sample rates in real time at playback?
When I play a video track with audio recorded at 44.1k or even an MP3...and playback on my soundcard set at 48k...it does play at correct pitch...even though theory of audio tells me that the sample rate is very inportant especially when playing back...sample rate needs to be the same on both the soundcard audio clock and the software...
Is this part of the magic of quicktime or am I missing something.
Even a bit more confused when I got a dialog file from a filmaker attached to his quicktime file that he thougnt was in 44.1k...was actually 48k...so when taken out of quicktime and put in another enviornment for editing that became really important and avident that the samplerate audio rules outside of quicktime applied...
Am I correct then in my understanding that when working only in the quicktime enviornment that the rules of audio sample rate do not apply?
Thanks for thoughts...
regards
vibes

All you may ever want to know about QT, audio and OS X:
http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudio/index.ht ml
QuickTime 7 is a major re-write and it now has many more options for audio. Multiple channel support (up to 24) and sample rates of double DV Stream settings.
Magic is how it works.

Need help: just upgraded to Mavericks, now can't use audio interface.

I just upgraded to mavericks, and now Itunes and my other programs don't recognize my audio interface.
It's a MobilePre USB preamp. I tried uninstalling and re-installing the driver without success. Interesting, QuickTime sends audio to the external preamp, the sound is filled with static. I've tried playing with latency, but that doesn't help.
I can't understand why Quicktime recognizes the external device, but Itunes doesn't. Suggestions are appreciated.
My machine is 21.5 inch late 2009 model with a 3.06 GHz Intel Core 2 Duo processor and 12 GB of RAM.

Hope this references the equipment you are using.
The last update to the drivers is 1.1.3 Nov 2012
http://avid.force.com/pkb/articles/en_US/download/MobilePre-Drivers?popup=true&N ewLang=en&DocType=1450&q=MobilePre
If this is correct perhaps an email to Avid could help. I would think that they need to update their drivers to suit Mavericks.
There is also a link on that page to an article about Pro Tools and state that it is not supported on Mavericks at this time.

Having issue with audio rendering

The original .mov file, when played back in quicktime AND in the viewer of final cut plays back smoothly and without error.
I bring it into the timeline, and I have to render the audio for playback, which I think is normal. However, when the render is completed, it doesn't take very long for the audio to start being REALLY strange. Remember how an old cassette player sounded when the batteries were dying? How the voice of who ever you were listening to would become really deep and slow? That's what the audio sounds like. What's up with that? Is there anyway to fix it?

when played back in quicktime AND in the viewer of final cut plays back smoothly.
Because FCP's Viewer and QuickTime Player simply play back what you actually have, regardless of its format.
I bring it into the timeline, and I have to render the audio for playback...
Now you are asking the audio to conform to the specifications of a particular codec as set up in Sequence > Settings. If it does not, you will have to render every time that a change is made.
As a rule, FCP expects digital audio to use a 48kHz sampling rate and be in AIFF format.
If yours came from an audio CD (44.1kHz) or an mp3 file, for example, it does not conform and might well produce unexpected results. Resample the audio to the correct specifications using Soundtrack, Compressor or QuickTime Player.

Using a USB2.0 Mixer with Audio Interface in GB '09 vs a A/I ?

Hello,
I am posting re: using a USB 2.0 Mixer (example Allen&Heath Zed Series with or without effects, Mackie USB mixers, etc, to use
rather than a Audio Interface (Which I have done in the past, though one I have did not keep drivers updated past OSX 10.5
Leopard, I have bought and returned a Firewire 400 and a USB 2.0 Audio Interface(s) in the past 10 months, so No recording
I can use a Podxt to get Guitar in via USB but not XLR mics (Vocal or Amp mic's) I have read and read and spoke with people
and get a variety of answers, I am inquiring is this a workable way to get both 1/4 HI-Z inputs (Guitar or Bass) XLR Mics
and line ins to the mixer and USB 2.0 to/from my iMac? I am just looking at going in a different direction, and using Garageband
the Audio Interfaces I had bought both had on screen mixer-routing (which drove me crazy and to me un needed)
Prior I used a bus powered Tascam US-122, I do want extrenal power (and both Audio Interfaces did that I bought and returned)
I just need a basic sound in-sound out etc, with decent preamps , I have Mics, Guitars, Monitors, MIDI Keyboard, etc
just lacking the source in, (I also wish to avoid headphone monitoring due to ear issues which I do not desire to further
aggrevate, I hope this makes sense,
Thanks,
diamondblues

foxyroxyD wrote:
digital piano - midi in/out - audio interface - usb - Macbook.
check the cabling:
http://www.bulletsandbones.com/GB/GBFAQ.html#controllertointerface
(Let the page FULLY load. The link to your answer is at the top of your screen)

Sample rate issues in Audition CC and CS6

I'm using a Blue Spark mic with the usb adapter and I keep getting an error message saying the sample rates don't match the device. I've tried several different sample rates and none of them work. I can monitor the audio from the mic but when I hit record the error message pops up.
Here is the message I keep getting and the settings I'm using: http://imgur.com/a/WpJST. (For some reason the insert image button isn't working so I just linked the screenshots through an image host.) For some reason the icicle doesn't show up as an output device so I just chose a default.

Try the Bob Howe's suggestion in this thread:
http://forums.adobe.com/thread/940613

Changing sample rates for recorded audio?

I've been trying to record VO through premiere. I'm using an m-audio mobile pre which has never given me trouble in any other application. When i turn on Meter Input(s) Only from the panel menu of the track mixer i can clearly see the input move in time and responsively. When I record however the resulting clip is much shorter than the ammount of time recorded and the audio is all garbled and chopped. It clearly jumps out to me as a sample rate problem. I looked at the clips and they are 44.1 but the sequence i'm recording to is 48. I made a 44.1 sequence and recorded to that which worked fine but there must be a way to change the recording sample rate right? I can't find a solution anywhere. Anybody know?

i'm just following up here. Thanks for the replies. I understood i would lose data as one answer suggests, but i wanted to convert it so i could work in a new song environment in which all the NEW tracks were recorded at 192. So the second answer was very helpful. incidentally, with further research we are recording at 96.

9.1.7 buggy with audio interface

So, my PreSonus Firebox regularly goes silent since the 9.1.7 update.
This seems to always happen if I put the computer to sleep and wake it again... sound becomes unavailable until I toggle audio output settings in Logic, or quit the program.
Output from any other application is also silenced. It's restored when I quit Logic.
Anyone else had this problem with their interface?
Thanks...

This problem actually has to do with more than waking-up issues. Sometimes the interface goes silent just when Logic is put in the background for awhile. Regarding the necessity for restarting-after-wake to keep things going, I can't recall ever having a problem similar to this since the days of OS9, and I work daily with Logic and many other audio apps. My firewire interfaces (Presonus and other) have been solidly reliable upon waking.

LogicX - Maverick - Sample Rate issue with Audio Interface

Similar Messages

Maybe you are looking for