MiniDV audio sample rate issue

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

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

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Audio sampling rate issue using USB 6008

Hello,
I'm trying to input audio into the analog ports of my DAQ 6008. I'm using labview 8.5, and generated the DAQ assistant (express), used array index to output the 4 microphone channels into the waveform chart IV (i.e. I'm display 4 charts for the four analog microphone inputs).
when I run it, the charts, on the screen, update every 1 sec it seems, which is MUCH slower than I would like to sample at (I perfer the max, of around 10K/s). does anyone know why it's sampling so slow, and how I can improve this?
thank you,
dave

i cannot use waveform graph instead of chart, becuase i'm gathering scalars, and waveform graph requires at least 1D array data. Also, in DAQ assistant, how do I set these blocks your refering too? The only sampling options i see are sampling type (i.e. continuous), samples (which doesn't matter for continuous sampling), and rate (which I've maxed out at 2.5k).
In the DAQ assistant, the rate of data is being displayed at the correct rate, but in the front panel, and in the file I'm writing too, I'm see only about 1-2 data poins per sec....
below is a copy of my VI (note, the system automatically insets a dynamic to array converted.
thank you!
dave
Attachments:
audiosensory.JPG ‏209 KB

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

Inconsistent Audio Sample Rate - Export issue

I am attempting to export my project into a Windows Media file and I'm getting an error that says "inconsistent audio sample rate" Does anyone know how to fix this problem?

That error was introduced with the release of Quicktime 7.2. Flip4Mac released a beta with the fix for that particular audio error thing HERE. I don't think the final is ready yet, just the beta, but it's working for me. Don't be "put off" by it, but, before you can download the beta, they ask you to fill out some registration info so you can report back any problems you might encounter.

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 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.

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

Audio Sample Rates Don't Match

I am capturing a MiniDV tape via Firewire to a Final Cut project
w/ a Capture Preset of DV/DVCPRO-NTSC w/ 2 channel 48KHz 16-bit
audio sample rate
and when I check the Item Props on said clip in Browser it says the same info
I've toggled both Mono and Stereo on Capture Settings, but after capturing,
in all cases, I get the same warning message:
Audio Sample rate one your captured media files doesn't match the sample rate on source tape. This may cause it to go out of sync. Make sure the capture preset matches sample rate of tape.
Please advise, thanks

I am capturing with a Sony MiniDV camera w/ a Firewire400 cable,
and the tape is a Sony MiniDV (DVM60) Premium tape
I am unsure of the camera that shot the footage, which is from 1998
and was one of the earliest MiniDV cameras (a large one)
I tried both Firewire Basic and Log and Capture
but the problem still persists
any other options?
thank you

Error Msg.: "Inconsistent Audio Sample Rate" with Flip4Mac

If you're using Flip4Mac to produce WMV files, the latest QT updates will create a conflict with Flip4Mac, which produces the error msg. referenced in the subj. heading above.
If you're using Flip4Mac, you may want to wait until Telestream resolves the issue before upgrading QT.
More info is here: http://www.flip4mac.com/knowledge/kb_0070.htm
and here: http://forum.flip4mac.com/forum/messageview.aspx?catid=10&threadid=270&enterthre ad=y
MacPro, MBP 17in (2 GB RAM), PB G4 15in, PB G4 12in, iMac G3 Mac OS X (10.4.9) Loyal to Apple since the Apple ][+

Just in from flip4mac support:
The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

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.

Why is FME audio sample rate restrictive?

I am unable to select higher than 22kHz audio sample rate on
the FME Control Panel - this results in very poor audio quality
delivered! The input device delivers up to 48kHz and is succesfully
addressed by WME - the only reason why I'm still using WME.
In case you are interested, I have listed the input device
spec below:
DVBETA (LairdTelemedia)
Output Ports available (I use FireWire)
SDI Digital Video: SMPTE 259, PAL or NTSC, on BNC a
connector
AES/EBU Digital Stereo Audio: 16 Bit @ 48 kHz, Embedded on
SDI connector
* IEEE-1394 (FireWire) x2: maximum 400 Mbps, PAL or NTSC, on
a 4-Pin and 6-Pin connectorsMachine Control, RS-422, on a 9-Pin
connector
Time Code: DVITC or VITC, on a BNC connector

I have the same issue when using my video capture card in FME
on Vista.
I am having a lot of issues actually.
1. CPU while encoding video & audio is only reaching
around 15% but I am dropping frames when I try and use high quality
encoding settings.
2. Audio and Video get out of sync very easily.
3. Audio sounds VERY odd, especially during music. It seems
like the speed of the audio is constantly shifting. Makes things
sound like oooOOOoooOOoo etc..
I hope Adobe is addressing these issues.
Thanks and take care.

Audio sample rate conversion

Hello again.
In my timeline I have audio on track 1 from a mini DV - sample rate 32kHz.
On tracks 3&4 is a stereo track from a CD at 44.1kHz.
If I try and play together them the audio on track 1 is mute. However if I disable the monitoring on 3&4 and then play the timeline I can hear track 1 fine.
I assume its a sample rate issue though I'm probably wrong.
Can anybody shed any light on this.
Many thanks

Use Soundtrack to convert all your non-compliant audio to 16 Bit 48kHz AIFF files.
Your problems will disappear.
Set up your camera to shoot 16 Bit 48kHz for next time.

Audio sample rates

I am getting the message after capturing "The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape......etc". The only thing I notice is in the Aud Format window the captured audio is "16-bit integer" and the sequence is "32-bit Floating Point". Should I ignore this or is there a way to get them to match?
G5 dual 2.7 Mac OS X (10.4.5)

Hi mmona,
I did a little bit of googling and reading about 32 bit floating and 16 bit integer, and I left a bit more puzzled and without enough knowledge to be dangerous.
However, the warning FCP is giving you is for the sample rate, as opposed to the audio format.
To see what I mean, highlight the clip and then hit Command-9. Audio rate and audio format are the 3rd and 2nd to last items. The warning is refering to the audio rate, but the difference you are reporting is in the audio format (which I remain curious about.)
I'm going to go out on a limb and guess that you are capturing DV that originated in a camcorder that cost less than $10K. In many cases, these camcorders capture unlocked audio. Google unlocked DV audio and you'll get a lot of hits, and many explainations.
I've gotten the warning you describe, but only with DV originating in certain less expensive cameras. Not all cameras are equal when it comes to this. The XL1, despite being a higher end camera in this class, often shows this issue.
Sometimes in these cases, when I look at the audio rate on the clip (3rd from bottom after hitting Command-9) it will report a number like 48.009. The practical effect is that the audio drifts slowly out of sync as the clip progresses in FCP's 48 KHz world.
So I keep my clips short and the problem remains invisible.
Tom

MiniDV audio sample rate issue

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