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

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Flash - finalcut - and audio sample rates

So, I have a complex issue I'm trying to work through in my head and I thought one or two of you brilliant people could provide some insight.
I've been provided with a 11min flash animation that had been turned into a 24fps mov. There were some issues with some garbage frames, so I brought it into finalcut to take out those little bits of the video. Now I'm trying to make this clip into a DVD friendly video format and a youtube friendly one. In the original clip there is a circle that fairly smoothly grows larger. When I brought it into FCP that part of the video had all sorts of jagged edges and messyness.
Is that from frame rate going from 24 -> 29.97 ? Is that keyframeing issues? What can I do to solve that?
second...
the audio track is at 48khz. Another guy took that track to go add music, clean up the audio and give it back to me. I got the file back at 48khz..brought it into FCP and placed it underneath the original track. They both start at the same time and then the new track gradually starts to drift and no longer be in sync.
What is up with that?
Any and all help will be loved, used, and enjoyed.

So...no thoughts on this? I can't seem to figure out why 2 recording are playing so far off each other.

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.

Error message trying to synchronise audio and midi sample rate 42463

I have a Rode Podcaster mic that is terrific. It has been working fine in GB. suddenly I am getting this when I try to record my podcast:
'Error message trying to synchronise audio and midi sample rate 42463 recognised. Check conflict between Garageband and external device'
Has anyone experienced this before? How do I resolve it? I'm still using my Version 6.0.5.
Apple, is this a way of forcing me to upgrade to V10?

Just click the icon on each track that stops it acting as a live input. It looks like a sound wave. Only have them clicked orange when using the track to record! I don't know if that will solve your problem!

Error while trying to sync audio and MIDI, sample rate 39100 recognised??

When I press record a message appears saying 'error while trying to syncronise audio and MIDI, sample rate 39100 recognised. Check conflict between garage band and external device. I'm using a Behringer UMC-202 interface. The inputs are selected as the device but no audio is present either.

Just click the icon on each track that stops it acting as a live input. It looks like a sound wave. Only have them clicked orange when using the track to record! I don't know if that will solve your problem!

Audio and visual sample rates

hello
everytime i capture using FCE it comes up with the warning that my rates do not match and says 'make sure the audio sample rate of your capture preset matches the sample rate of your tape'.
when i ignore this warning, the visual and audio often match nonetheless. then at times they do not, and there is a definite lag.
i posted a question about this before and the reply was to change the settings in Easy Setup to DV-NTSC. however, this did not work. i bought my camera in the uk (it is a sony DCR PC330E PAL) and so can't really see why ntsc would work.
im sure this is a simple problem, but i am too inept to solve it! can anyone offer any help?
thanks in advance

Hi(Bonjour)!
It's important to match video and audio format between your sequence(project) and your shooted material(camcorder settings).
If you camera support PAL or NTSC, select the correct video format in easy setup dialog. That's the video format.
Check your camera settings for audio format: there are 2 options 12 or 16 bit (in standard DV). It's preferable to use 16 bit with FCE.
Select the correct audio format in easy setup dialog: if your shooted in 12 bit, select 32 kHz in easy setup. If you shooted in 16 bit, select 48 Khz in easy setup. That's the audio format
Michel Boissonneault

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.

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?
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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

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!

How is Core Audio sample rate set?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

You might ask on the GarageBand, Logic or Final Cut forums. There is not much audio traffic here.

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

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