Core Audio - Sample rate

I've just got myself a copy of Logic Pro 8 as a complete newbie and have hit a bit of a hurdle within 24 hours of opening the software.
I was using the software without a problem just tinkering around using a Carillion MIDI interface to play some software instruments. I don't have a mic yet so thought I'd try my bluetooth headset just so I could play with recorded vocals. When I started speaking while Logoc was recording it came up with an error message
"Core Audio
Sample Rate 8000 not allowed."
I then turned off my bluetooth headset and turned bluetooth off on my Mac so I could go back to using the MIDI keyboard but got no sound. I closed Logic and re-opened, selected "Empty Project" withing the "Explore" tab and was again presented with a series of the messages, all reading as per the above error messages but I get 3 with "8000" and one with "73536".
When I'm playing the keyboard I can see the note I'm playing or the chord in the dialog box at the bottom of the Logic interface screen but again, no sound.
Any help would be hugely appreciated. Again, I am a COMPLETE Logic novice so please be gentle with me.
Regards,
Mark

Hi Jounik,
I went into Audio MIDI Setup and the headset didn’t appear. The keyboard appears as “Midilink” but I couldn’t see where to check the sample rate. I did however use the Test Setup facility and the keyboard was only producing a sound when the key was released.
I also had a look at: Logic Pro > Preferences > Audio (Devices) but wasn’t really sure what to look for. This is what I was confronted with http://adoseof.co.uk/resources/Picture1.png
The project sample rate was set to 44.100 KHz
I like the idea of using the internal mic. I’d tried that before but was getting feedback. Plugging the headphones in seems like a good solution.
To try and get the keyboard working again I opened a new Empty Project and left the dialog with the default settings as indicated here: http://adoseof.co.uk/resources/Picture2.png
I’m not sure what you mean by “In the I/O slot of the channel strip open a synth, e.g. ES1”
I then used the on-screen keyboard and still no sound was coming out.
Any other ideas?
I tried the keyboard in garage band and it worked fine so I can only assume it must be something to do with Logic.
Mark
Message was edited by: hotsawz

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Interval Since Last Panic Report: 2159 sec
Panics Since Last Report:          1
Anonymous UUID:                    B03C4E76-CB50-4839-8FB4-A6A511E50492
Fri Sep 23 20:10:24 2011
panic(cpu 0 caller 0x001AB0FE): Kernel trap at 0x00d67395, type 14=page fault, registers:
CR0: 0x80010033, CR2: 0xb795e750, CR3: 0x011a4000, CR4: 0x00000660
EAX: 0x00000000, EBX: 0x00000004, ECX: 0x00000040, EDX: 0x00000000
CR2: 0xb795e750, EBP: 0x2e76fbd8, ESI: 0xe00007c8, EDI: 0x3795c830
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0x2e76f9e8 : 0x12b4c6 (0x45f91c 0x2e76fa1c 0x13355c 0x0)
0x2e76fa38 : 0x1ab0fe (0x469a98 0xd67395 0xe 0x469248)
0x2e76fb18 : 0x1a1713 (0x2e76fb30 0x2e76fc4c 0x2e76fbd8 0xd67395)
0x2e76fb28 : 0xd67395 (0xe 0x48 0x10 0x10)
0x2e76fbd8 : 0xd65b1b (0x37a70000 0x3795c830 0x4 0x10)
0x2e76fc58 : 0xd519be (0x4678000 0x37b00bc0 0x37a70000 0x22f)
0x2e76fc88 : 0xd51bcb (0x4678000 0x37b00bc0 0x37a70000 0x22f)
0x2e76fcd8 : 0xb528a0 (0x4678000 0x22f 0x100 0x41cfb68)
0x2e76fd28 : 0xb52687 (0x41cfb00 0x22f 0x100 0x0)
0x2e76fd68 : 0xb52463 (0x41cfb00 0x3782c010 0x37af8000 0x22f)
0x2e76fde8 : 0xb4db00 (0x41cfb00 0x47c5908 0x22f 0x1)
0x2e76fe68 : 0xb4d18a (0x4371200 0x22f 0x1 0x455d580)
0x2e76fea8 : 0x4437c8 (0x4371200 0x22f 0x1 0x0)
0x2e76ff08 : 0x198fa3 (0x2e76ff44 0x0 0x0 0x0)
0x2e76ffc8 : 0x1a1cfa (0x533c6c0 0x1 0x10 0x0)
No mapping exists for frame pointer
Backtrace terminated-invalid frame pointer 0xb035fb58
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         com.m-audio.driver.firewire.dice(2.3.2)@0xd4d000->0xd77fff
            dependency: com.apple.iokit.IOFireWireFamily(3.4.9)@0x795000
            dependency: com.apple.iokit.IOAudioFamily(1.6.9fc5)@0xb43000
         com.apple.iokit.IOAudioFamily(1.6.9fc5)@0xb43000->0xb59fff
            dependency: com.apple.kext.OSvKernDSPLib(1.1)@0xb40000
BSD process name corresponding to current thread: Logic Pro
Mac OS version:
9L31a
Kernel version:
Darwin Kernel Version 9.8.0: Wed Jul 15 16:55:01 PDT 2009; root:xnu-1228.15.4~1/RELEASE_I386
System model name: MacPro3,1 (Mac-F42C88C8)
System uptime in nanoseconds: 1278160883646
unloaded kexts:
com.apple.driver.InternalModemSupport          2.4.0 - last unloaded 190309720723
loaded kexts:
com.m-audio.driver.firewire.dice          2.3.2
com.apple.driver.InternalModemSupport          2.4.0 - last loaded 9798699115
com.apple.iokit.IOBluetoothSerialManager          2.1.9f10
com.apple.filesystems.autofs          2.0.2
com.apple.driver.AppleHDAPlatformDriver          1.7.1a2
com.apple.driver.AppleHDAHardwareConfigDriver          1.7.1a2
com.apple.driver.AppleUpstreamUserClient          2.7.5
com.apple.driver.AppleHDA          1.7.1a2
com.apple.kext.ATY_Lamna          5.4.8
com.apple.driver.AppleHWSensor          1.9d0
com.apple.Dont_Steal_Mac_OS_X          6.0.3
com.apple.driver.AppleTyMCEDriver          1.0.0d28
com.apple.driver.AppleHDAController          1.7.1a2
com.apple.iokit.IOFireWireIP          1.7.7
com.apple.driver.AppleUSBDisplays          2.0.2
com.apple.driver.AudioIPCDriver          1.0.6
com.apple.ATIRadeonX2000          5.4.8
com.apple.driver.AppleMCEDriver          1.1.7
com.apple.driver.ACPI_SMC_PlatformPlugin          3.4.0a17
com.apple.driver.AppleLPC          1.3.1
com.apple.driver.CSRUSBBluetoothHCIController          2.1.9f10
com.apple.driver.AppleHIDKeyboard          1.0.9b4
com.apple.driver.AppleUSBMergeNub          3.5.2
com.apple.driver.CSRHIDTransitionDriver          2.1.9f10
com.apple.driver.PioneerSuperDrive          2.0.9
com.apple.iokit.SCSITaskUserClient          2.1.1
com.apple.iokit.IOATAPIProtocolTransport          1.5.3
com.apple.driver.XsanFilter          2.7.91
com.apple.iokit.IOAHCIBlockStorage          1.2.2
com.apple.driver.AppleFileSystemDriver          1.1.0
com.apple.driver.AppleUSBHub          3.4.9
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com.apple.driver.AppleIntelPIIXATA          2.0.1
com.apple.driver.AirPortBrcm43xx          367.91.22
com.apple.iokit.IOUSBUserClient          3.5.2
com.apple.driver.AppleIntel8254XEthernet          2.1.2b1
com.apple.driver.AppleFWOHCI          3.9.7
com.apple.driver.AppleUSBEHCI          3.4.6
com.apple.driver.AppleUSBUHCI          3.5.2
com.apple.driver.AppleEFINVRAM          1.2.0
com.apple.driver.AppleACPIButtons          1.2.5
com.apple.driver.AppleRTC          1.2.3
com.apple.driver.AppleHPET          1.4
com.apple.driver.AppleACPIPCI          1.2.5
com.apple.driver.AppleSMBIOS          1.4
com.apple.driver.AppleACPIEC          1.2.5
com.apple.driver.AppleAPIC          1.4
com.apple.security.seatbelt          107.12
com.apple.nke.applicationfirewall          1.8.77
com.apple.security.TMSafetyNet          3
com.apple.driver.AppleIntelCPUPowerManagement          76.2.0
com.apple.driver.DiskImages          199
com.apple.BootCache          30.4
com.apple.iokit.IOSerialFamily          9.4
com.apple.driver.DspFuncLib          1.7.1a2
com.apple.iokit.IOHDAFamily          1.7.1a2
com.apple.iokit.IOAudioFamily          1.6.9fc5
com.apple.kext.OSvKernDSPLib          1.1
com.apple.iokit.IONDRVSupport          1.7.3
com.apple.iokit.IOGraphicsFamily          1.7.3
com.apple.driver.IOPlatformPluginFamily          3.4.0a17
com.apple.driver.AppleSMC          2.3.1d1
com.apple.driver.AppleUSBBluetoothHCIController          2.1.9f10
com.apple.iokit.IOBluetoothFamily          2.1.9f10
com.apple.iokit.IOUSBHIDDriver          3.4.6
com.apple.driver.AppleUSBComposite          3.2.0
com.apple.iokit.IOSCSIMultimediaCommandsDevice          2.1.1
com.apple.iokit.IOSCSIBlockCommandsDevice          2.1.1
com.apple.iokit.IOBDStorageFamily          1.5
com.apple.iokit.IODVDStorageFamily          1.5
com.apple.iokit.IOCDStorageFamily          1.5
com.apple.iokit.IOSCSIArchitectureModelFamily          2.1.1
com.apple.iokit.IOAHCIFamily          1.5.0
com.apple.iokit.IOATAFamily          2.0.1
com.apple.iokit.IO80211Family          216.1
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com.apple.driver.AppleEFIRuntime          1.2.0
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What is the best way to deal with different audio sample rates on the same timeline ?

what is the best way to deal with different audio sample rates on the same timeline ?

You don't have to do anything special. If possible, start your project with a clip that has the desired target frame rate and audio sample rate, and your project parameters will be set automatically. Other sample rates will be converted under the covers.
For example, if your video is shot at 48khz, you can add music files at 44.1khz with no problem.
If you are recording audio that you want to synch with video (multicam), you will get best results if everything is 48khz, but you can use 44.1 if that is all you have. Once I forgot to reset my Zoom to 48,000 and it still worked.

How do I change the audio sample rate from 48kHz to 44.1kHz for Mpeg2

Hey all,
I've been searching for a while but haven't found any direct answers in the forums or the user manual so here goes.
I have to dispatch a file to Bloomberg TV and the file specs they have given me are as follows:
Video Standard: MPEG-2, MP:ML, 4:2:0
Frame Rate: PAL 25fps
Video Size 720 x 405
Aspect Ratio 16:9
Audio Standard MPEG-1 Layer 2
MPEG-2 Program Stream Mux rate 6mbs per second
Bit Rate Type: CBR
Video Bit Rate 5.7mbs
Audio Bit Rate 192kbs
Audio Sample Rate 44.1Khz
Interlacing: Upper Field first (why they want interlaced for web streaming is beyond me)
GOP Structure: IBBP
I-Frame distance 12
Now everything above is fine except the audio encoding because even though I have set up a new setting from scratch I cant find anywhere to adjust the audio sample rate. The Inspector tells me the Audio encoder is set to:
Format: MPEG
Sample Rate: 48.000kHz
Channels: 2
Bits Per Sample: 16
Anyone Know how or even if I can change these audio settings? The only adjustments I can find are the filters or the transport/program stream option. I have it set to program as specified by Bloomberg.
Thanks in advance
J

The only setting that I could find in compressor that lets your change the bitrate to 44.1 is when you create a new dolby digital setting and then under the inspectors audio tab/Target System button, change the button to Generic AC-3. When done, you can change the Sample Rate to 44.1.
Hope this helps?

Problem with audio sample rate when exporting

I am having an issue with my audio sample rate. For some reason it is set at 8 khz by defualt when I am exporting. This was not an issue until the latest update (2 weeks ago). I'll try and give all the info I can up front. Also I tried customer support...that was a nightmare.
Ok the issue I am addressing is that the audio sample rate when exporting is at 8khz by default. Which to my knowledge has NO relevant use what so ever. Even when I set the file format to h.264 it insists on resetting to 8khz. Even if I manually set everything look at the render queue and click on output module again it changes the sample rate back to 8 khz automatically which I must adjust again. Why can it not just be set at 48000 like every other adobe product is and like it used to be two weeks ago. Does anybody have an answer?

This problem is addressed by the After Effects CC (12.2.1) bug-fix update, which is now available:
http://adobe.ly/AE_CC_1221
Note the part at the end of that page about a crucial update for the Creative Cloud desktop application, which addresses some severe problems with AME, Premiere Pro, and After Effects.

Highest audio sampling rate in CS4?

Hello,
I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
What is the highest audio sampling rate that can be utilized in Premiere Pro CS4? Can it import and export 192kHz 24-bit audio?
Thanks in advance

Hey Hacienda,
I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so. But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry. So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive). But we do share the neophyte status when it comes to video editing :-P
Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe. It is a theorem, meaning it's already mathematically proven. It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it. To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies. Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz. You need to spend BIG dolars for anything that goes beyond that. So, who are we really making music for here? The super rich? Dolphins?
Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform. OK, well, this is the kind of snake oil marketing BS I was talking about. The biggest one being that 1bit DSD crap that Sony/Phillips is pushing. Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound. It'll just make the files bigger for no reason. Again, the Nyquist Theorem already proves this. This is FACT! Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
Don't forget that modern converters already sample at much higher frequencies than the target sampling rate. I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate. But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform. Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters. These components are what make a converter high grade, among others. The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
By the way, I didn't say humans can only hear up to 20KHz. I'm sure there are people who can hear above that. My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that). Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details). However, I do agree with you as far as compressed audio goes. Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper. Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net. But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats. As far as digital media distribution goes, it's the future and companies are seeing that. More and more people download music rather than buying CDs, so I do believe those numbers are accurate. Just look at sales from iTunes and even games like Guitar Hero and Rock Band. It's just a matter of time.
Take care!

DV 16:9 but Only Exports 4:3 WHY? Also Audio Sample Rate Problem

I'm quite new to Final Cut and have FC 6.01. I use PAL 25fps and a 16:9 SD DV Camcorder with FireWire SONY DV VTR Deck. I have two problems:
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2.(Not sure if its related) I get the following message everytime after I capture individual clips or if i press the Escape key during a capture:
"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 media files to be out of sync. Make sure the audio sample rate of your captured preset matches the sample rate of your tape"
Does any one else have this problem?
Apple, as yet have not given me any answers.

Danny Boy.. Thanks for your reply and I'll be happy if it is my fault and not FCP's. Actually i made an error in my post. It does indeed display correctly in DVDSP, it's iDVD it does not even when told to display it in 16:9. However, when i used the PAL DVD Anamorphic file preset, iDVD still couldn't display it. To get it working I had to tell Compressor specifically (in the additional settings) to encode 16:9 despite what the presets stated! No matter how one looks at this, this is confusing to say the least! If a preset says 16:9 then one should expect 16:9! Remember, Im using the display window of iDVD to show me the output.
To detail my steps as hanumang has said, im doing the following:
1. in FCP I encode to QuickTime Move
2. Open iDVD & Create a Project
3. 'Drag' The QT file into the menu
4. Using iDVD Preview function, Preview the QT file
5. ITS STILL 4:3
now, the above was done with QT Conversion which as also set to encode 16:9 and still had the same result.
Thnaks to you both.

Change capture settings to 32k audio sample rate???

I have a DVCAM tape that my client recorded and the audio sample rate is 32k. I have noticed that the audio seems to get out of sync when I capture it using the 48k NTSC. I also get a warning saying that my source tape's audio sample rate doesn't match the capture settings and that audio could be out of sync. I looked at past posts and it does seem to be the 32k verses 48k.
My question is how do I change my capture settings to 32k? I see how you do it on the sequence settings, but I can't figure it out on the capture settings.
Thanks,
John Schenkel

John,
I recently read a post where there were four likely answers to the problem, yet I was not sure which one was the best solution.
Why? Because even though the question was marked as answered the person posting the original question didn't mark which individual's answer was the solution.
Please help future users to find answers more quickly, who knows, you may find yourself in the same situation someday.

Capturing issues-audio sample rate & locating timecode break

I am a first time FCP user.
I am working on an educational video with hours of DV source tapes.
The source tapes are from a 2R50MC Cannon mini DV camcorder also used as the deck I am capturing from via firewire. My scratch disk is Maxtor III 500GB external drive hooked up via a firewire (I'm realizing the firewire is years old could that be a problem?)
In my first capture test I captured 1 minute of tape by setting in and out points and after completeing the capture I received an error message—"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 media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape."
For the past day I have been scouring the user manual, this discussion group and training tapes from Lynda.com and can not resolve this issue. I have determined that the source tape is 12 bit or 32 kHz and yet can not find a way to set up capture preferences to 32 kHz. Is this the problem/resolution? Page 320 of the user manual shows the QuickTime Audio Settings dialog box but I can't find it in FCP or Quicktime. Is this where I make the change to 32kHz?
After doing the first test, I tried again and ran into a new problem—it stopped capturing with the message: Locating timecode break [press esc to abort]. It never seemed to resume capturing and I pressed escape to abort. If I'm reading FCP time code window correctly I received this time code break message within a section I had already captured previously (with the audio sync warning).
I have done several capture tests at different points in the tape and on different tapes all with the same results. I've used the capture clip and capture now buttons. I've tested with drop frame turned on and off. Confirmed my setting of: At timecode break "Make New Clip." Confirmed my Easy Setup as DV-NTSC. My capture preset is DV NTSC 48 kHz. I've turned off and on FCP. Restarted my computer. Restarted my computer with the shift key down and ran permissions.
Any ideas are GREATLY appreciated, T.

I am concerned about available memory when capturing to my hard drive and the babysitting and extra steps involved considering the amount of tape I want to capture...but it does work that's GREAT!
This is always a concern, but in your case I think having the camera and external on the same bus was causing your problem. You may have to capture a little and then transfer, rinse and repeat. Just don't try to do too much at once and let your System Drive get too full, you'll run into other problems there. Slow and steady is the pace!
I shouldn't have stacked my questions since you answered one and Chris answered the other. Don't know how to apply the answered question and who gets the points.
No problem, just mark it answered and divide up the solved and helpful points as you wish. All in all we really don't care too much about the points, but they do make us feel good! Thanks for your desire to use the forums properly.
K

How do you set sequence audio sample rate?

I tried posting this to another, but it was already answered, and noone will see it.
I am getting the capture error "audio sample rate doesn't match" and yes, I can see in my browser that the clip is 48khz/16bit, but the sequence is 48khz/32 bit. Howver, wherever I look to change the sequence setting, it is 48/16 already. I've gone to FCP on the menu dropdown to audio/video settings - it's correct all through there. I've gone to the menu dropdown Sequence settings, and it's correct there. I've closed down, opened a new sequence, restarted, everything I can think of. Is there a secret to getting them to match? And, can I fix a project already edited with this discrepancy? Its export to QT is WAY out of synch.

Annoying - I can't see your post when I am in reply mode.
Yes, I get this error when i am capturing. From reading a previous post about the error, I thought the solution was to check the audio rate of the captured clip in the browser, and then make sure it matched the audio rate of the sequence. Like I said, everywhere you get to change the sequence setting, it SAYS it is 48/16, but yet, whe I scroll over in the browser, it says the audio rate is 48 KHz and the audio format is 32-bit floating point. Am I looking at the right places?
I can't check the settings in the camera until this evening...don't ask.
I'm not so sure this is not a QT issue instead of anything to do with capturing, etc. It plays back fine in the timeline.

Capturing-audio sample rate

I have fce hd v 3.5. My camera is a sony dcr-trv11. I am trying to "capture now" but it looks like I am dropping frames. When i hit "esc" it stops capture of course, but I get a message that reads "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 ans audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape." Okay, so how do I do that? When I originally captured my footage with imovie I had no problems. I had trouble importing them to FCE so I scrapped them all and decided to start from scratch with FCE. I have 199.5 GB of free space.

Most miniDV cameras can be set to record audio as 12-bit or 16-bit; and most of them come set to 12-bit as the default.
Your preset (easy setup) in FCE has to match both the video & audio setup of your camera. Normally, you would select the DV-NTSC easy setup in FCE, which would give you a sequence that expects DV-NTSC video and 16-bit (aka 48KHz) audio. If your video was recorded as 12-bit (aka 32KHz) audio, but you captured into a 16-bit (48KHz) sequence in FCE, that would give you the mismatch.
Check your camera - if it was set to 12 bit audio for the tape you are trying to capture, then in FCE you should select the DV-NTSC 32KHz easy setup for your sequence before you capture your tape.
There are many different reasons you might be getting dropped frames - can you tell us more about your exact setup, esp. if you have an external HD connected to your system. Oh, and by the way, 512MB is the bare minimum to run FCE, you will find things much better overall if you upgrade to at least 1GB.

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.

Core Audio - Sample rate

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