Play audio samples in different rates

I need to play samples from differents rate, I'm working about speech and most of the time, the rate of the file is 16000Hz, the sound VI from National instrument don't allow a rate different from 8000, 11025, 22050 and 44100...
Maybe, It could be possible to change the DLL "lvsound.dll" include by National instrument but I haven't the source code.
I also have the same problem for recording speech.
Thanks for your help

Under windows 98 or 2000, you can do the following to play the rates not supported by the NI lvsound.dll. Use activX container, select Windows Media Player (v 6.4 and above). Pass the complete path of the wavefile you want to play to the property node and enable the play action. Oh, you can not only play sound files but also media files supported by the WMP. The NI's write to sound vi can be modified to write wave files with any rates.
I haven't tried the recording side so I don't know it works or not.
Hope this helps you.
Joe
Roush Industries, Inc

Similar Messages

Application that plays audio file at double rate

I need to transcribe audio files and need an application that will play audio files at double their rate (and half their rate) so that I can quickly go through audio files and then concentrate on the pertinent portions. Any suggestions of applications for Mac OS X? Free is always good!

Dear Matt,
Thank you for your suggestion. I checked it out. As I am transcribing portions of the files, I would like to just change the playback speed, not the actual file. It seems like, in Audacity, you actually transform the file. I noticed to do this you use "change tempo" to avoid pitch changes. However, by modifying the file, the file becomes longer or shorter and then I cannot reference a particular part of the file by a time (minute:second).
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How to resolve audio sampling and bit rate differences before sound mix

I'm editing a project where an additional recorder (Zoom H4n) was used and synched to the camcorder track with Plural Eyes. Some of the sampling and bit rates don't match within the Plural Eyes synchronized clips and I need to send this out for a sound mix. Could I use Compressor to transcode the audio? Would I have to go back to my original separate tracks, transcode in Compressor and then re-synch using Plural Eyes? Next time ProRes from the beginning -
but if there's an easier fix for my current problem it would be much appreciated.

Thanks Michael -- you've helped me with the project before (back in May of this year).
I started this project as a newby back in 2010 and was a bit overwhelmed (i.e. just happy to get everything into FCP and be able to edit.)
I'll try the media manage solution, but if that doesn't work I think I'll live with the one-frame drift. I'm assuming
the only other alternative would be to go back to all the original video and audio files; transcode using Compressor, and then re-synch with Plural Eyes and go back in to match up the edits?
Thanks to your continuing help, I should know how to set things up correctly from the get go for the next one!

Capturing audio & video at different rates

I have a Panasonic PV-GS700 camcorder, and use primarily Sony DVC Premium mini-DV tapes. For as long as I can remember, when I capture a long section of tape (maybe the entire hour), I wind up with a clip whose audio and video are at slightly different speeds, getting more and more out of sync the longer the clip.
I spend a ridiculous amount of my editing time meticulously adjusting the speed of the video, and it doesn't always seem to be off by the same factor.
My capture settings are usually Firewire NTSC / DV NTSC 48 kHz but sometimes I have changed to "Noncontrollable Device" if I am capturing an entire tape and I don't want to get hung up on capture errors. The firewire cable is normally routed through a firewire hub, but I have also observed the same problem when it is connected directly to the computer.
I'm curious:
1. if there is some setting I can change to fix this problem with audio & video capturing at different speeds?
2. is it likely to be a problem with my camcorder or the software? Are there decks that just handle playback of mini-DV tapes and work reliably with Final Cut Pro?

The camera says 16-bit in playback.
When I'm playing the tape from the camera and watching the viewscreen, the audio & video are in sync; it's only the captured DV file that runs at different speeds.
At this point, I'm considering a newer camcorder which shoots in HD and saves to a hard drive or SD card, such as a Canon VIXIA HF200. Would such a audio/video sync problem even be possible with that sort of camcorder?
(PS: If you were anyone other than Tom Wolsky, I would definitely be dubious that there is any part of my camcorder called a "timing crystal"... it sounds like cheezy sci-fi or Napoleon Dynamite )

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.

Importing audio - sample rate/bit depth

Hi forum,
I am working on project at 44.1K, 24 bit. Audio elements are being sent to me to be added. Some have come in incorrectly; at 48K, 16 bit. I can convert easily but didn't think I needed to.
I thought by selecting the "Convert Audio Sample Rate When Importing" option when creating the project that would all be worked out.
That is what I've done - and the file seems to be the correct pitch -- yet shows up in the audio window with it's original specs (48/16). Also ... will logic keep it at 16 bit and play all other files at 24 bit?
I want to be sure about this ... something seems fishy.
Cheers
Dee, Ottawa

Hi,
I am re posting.
Regarding the same project: must everything in the arrange window of a project be the same sample rate (and bit depth). My understanding is that there is real- time conversion during playback. That all file types supported by logic - and all virtual instrument samples are converted in real time to conform to the selected bit depth and sample rate of project.
I ask only as I recieved reference sound files to temporarily place in a mix to see how mix will sit when going to Post. Two audio files are almost a semi tone higher than they are supposed to be (which is odd - so I am pretty sure it was just quickly sung in the wrong key at there end). And one file which was supposed to be timed out is not lining up.
I can work around this on this project. And I can simply convert in the sample editor and re import to compare.
But again ... I just want to check my understanding for future reference. The manual indicates differing rates etc should not be a problem. (ie: That Logic allows one to have differing rates and bit depths. Inversely, the M Sitter video implies just the opposite.
I just want to be sure of this for future reference. Any adive?
Thanks in advance.
Cheers
Dee

Matching Audio Sample Rates In FCP

When the message comes up, "Audio sample rate must match sample rate of source tape (Log and Capture window) what exactly does one have to do to ensure a change to the correct .. ?

What I would really like first is for someone to point out the steps on the DSR11 remote for selection of both video and audio. I'm having to remember the three different selection places as I'm doing it "blind" and I can't see the results of my inputs whilst enabling it. The only confirmation I get is the Capture screen showing the SVHS tape playing if I get it right but that doesn't answer the question of how come I don't get a TV monitor screen showing up when then playing the captured material. That, as well as no audio which could possibly be due to a wrong audio preset in the DSR 11 due to my lack of visual. The video setting has to be right for me to get a Capture screen on my desktop. Normally, when starting FCP, the TV Monitor screen changes to a colour indicating connection.
With frustration at a peak, and just in case I'm in some way unhappily missing the right Capture preset, I'll bow to anyones step by step workthrough on the procedure.
Firewire between Macpro and DSR. SVHS in and out connections between DSR and SVHS VCR. Ongoing scart connection from VCR to TV monitor.

Audio sample rate does not match (HDcam to dvcam)

I'm trying to import clips and keep getting this 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."
Footage was originally shot on HDCAM and transferred to DVCAM elsewhere. Using FCP 5, am importing via firewire from a Sony DSR-11 deck. Using DV NTSC 48kHz Anamorphic as capture settings (though I've tried everything that I thought might possibly work with no success). The audio does not seem to drift over the course of several 5 minute or so clips. Clip settings show audio at 48 kHz (don't know if that's from capture settings or from actual data). Seems to me all audio should be 48 kHz 16 bits, so can't figure out what's going on. I have to export an EDL for the project to be finished in HD. Read some similar threads that ended in December, seemingly without much resolution. My broader concern is why this is happening; my immediate concern is do I need to worry about this right now since the media files will need to be recaptured in HD anyway. Any thoughts?
Thanks

A little more info. I'm having this problem on 4 tapes (from different cameras) that were transferred to DVCAM in a squished format to appear full screen on a 4x3 monitor. Video that was letterboxed and I can bring in with the standard DV NTSC capture settings does not have this problem. Still have the problem if I try to import the clips from the squished video with standard settings. Any thoughts?

Need help. Mac OS 10.5.8 crashes when changing audio sample rate.

Anyone have an answer to this problem?
Profire 2626 interface, when switching audio sample rate, crashes Mac Os

Usually when I disconnect the audio interface from my computer and then open up a song that has a different sample rate, the crash doesn't happen. But when the interface is hooked up (this problem started just a few weeks ago...), I get this error, pls bear with me, its long:
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
EFL: 0x00010202, EIP: 0x00d67395, CS: 0x00000008, DS: 0x00000010
Error code: 0x00000002
Backtrace (CPU 0), Frame : Return Address (4 potential args on stack)
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
      Kernel loadable modules in backtrace (with dependencies):
         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
com.apple.driver.AppleAHCIPort          1.7.0
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
com.apple.iokit.IONetworkingFamily          1.6.1
com.apple.iokit.IOFireWireFamily          3.4.9
com.apple.iokit.IOUSBFamily          3.5.2
com.apple.driver.AppleEFIRuntime          1.2.0
com.apple.iokit.IOSMBusFamily          1.1
com.apple.iokit.IOStorageFamily          1.5.6
com.apple.iokit.IOHIDFamily          1.5.5
com.apple.driver.AppleACPIPlatform          1.2.5
com.apple.iokit.IOPCIFamily          2.6
com.apple.iokit.IOACPIFamily          1.2.0

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!

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.

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

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

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

Mismatching Audio Sample Rates Upon Capture

I recently bought a cheap MiniDv JVC video camera, and I am having trouble capturing video and audio together in FCP 6.0. The capture seems fine, but when I finish an error pops up which reads: "The Audio Sample rate in your presets does not match the one on your source tape." As a result there is a delay between the video and audio tracks that creates huge amounts of extra work.
My video camera captures sound in 12-bit format, and from what I understand that usually corresponds to a 32MHz sample rate. I can't find any 32 MHz presets in the Easy Set Ups or other menus, the system seems geared entirely towards 48 MHz sound. I've messed around with some different combinations of settings, but so far to no avail. Any help someone can give me will be much appreciated.

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

Play audio samples in different rates

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