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

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

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

Message: inconsistent audio sample rate

when trying to Share my movie to create a .wmv file, I get the following message: "inconsistent audio sample rate -- the media you are exporting contains audio with multiple sample rates."
The audio to this clip is not the native audio. I replaced it ( on the second track - the first track audio I muted by turning off the check mark next to that track) with audio imported from Itunes. That audio clip is an aac audio file, 128 kbps, 44.1 khz. In expert settings I matched this, and indicated coding method: One pass constant bit rate (CBR).
Anyone know what it is I am doing wrong? Many thanks

No. That is not how it works.
44100Hz is the frequency and the audio sample rate could be 8, 12, 16 or even 24. Same with 48, 96 and higher frequencies. Open the Audio MIDI Setup app to see what your machine and sound card can offer.
QuickTime files can combine up to 99 tracks so a mix of sample rates wouldn't be much of an issue. WMP exports (I've never made one) seem to require only one audio source or constant sample sizes.
iMovie uses 48KHz because that is how it comes off DV tape. AIFF files are usually the same as audio CD (44.1KHz and 16 bit sample size) and I don't know if iMovie upsamples or otherwise changes them at import but I doubt it.
But 44100 does not equal 12 bit sample size.
Hope this helps clear that issue up.

Fixed but still says Inconsistent Audio Sample Rates

I imported some purchased music .aif files; completed my piece then tried to export a .wmv file through Quicktime. It told me I had "Inconsistent Audio Sample Rates" so I went back and re-exported the file through Quicktime with a sample rate matching the rest of my project.
I'm still getting the "Inconsistent Audio Sample Rates".
What do I need to do to get the project exported?
Cheers,
D

I have also been getting this error - just starting today. I tried exporting a mixdown in aiff, then converting that to MP3, bringing it back into a new sequence -- all with no luck.
However, I then exported this mixed-down version as a self-contained movie. Then, I did the wmv conversion with Cleaner 6 and it works. Slight glitch in frame 1 for some reason (motion file within the FCP sequence) -- but it seems like it's on the right track.
I also get the feeling this is due to the audio update. These are 8-track sequences I've been working with on and off for two months -- and the first time I try to render out after the audio, osx, and QT update, and it's crapping out on me. Could also be the changes to the h.264?
I've been spending all day on this, and I'm at my wits end!

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?

How can I find out the audio sampling rate of BetacamSP tape?

Hi guys
I'm trying to digitize BetacamSP tape. But I'm afraid if I might choose wrong setting...
This tape is from very long time ago so we don't know which audio sampling rate we recorded with..
How can I find out the audio sampling rate of this BetacamSP tape?
Thanks:)

The sampling rate is set by the Sony DVMCDA2 you are using, when the conversion is made from the analog input to the digital (DV) output. You should be outputting standard DV which is 16bit 48khz audio.
Assuming you are in the US, your Easy Setup for FCP should be DV-NTSC, and then open the Log and Capture Pane and set the Capture Settings Device Control to Non-Controlable Device and you should be good to go.
You will have to roll the deck manually and start and end your capture manually.
You can download a user manual for the DVMCDA2 by clicking here.
MtD

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

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:
1. When i capture my 16:9 DV footage the Logging Window shows only 4:3 with a distorted image in it. Though I have chosen the easy set up and told Final Cut that im editing Anamorphic 16:9 I also found that FC will only export 4:3. However, during the editing process i see 16:9 in the Browser Preview and Canvas Window.
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

Mixing audio sample rates

Similar Messages

Maybe you are looking for