Audio TC Rate woes -- importing no-TC wav as 25fps instead of 30fps

I am attempting to import audio clips into an old FCP 6.0.1/2 project from August on my work computer and even though the Easy Setup is 25fps base and the audio doesn't have TC stream, it keeps coming in as 30fps.
this is a real problem for long takes and pops up with a greenbar in timeline!
How can I stamp these audio files with 25fps code? Is there an external tool I can use to post synch vid and audio files and stamp audio files with aux TC and TC Rate?
Thanks!
matt

I got help for this issue over at CreativeCow's FCP pages -- it turns out, all I had to do was open a new project using an easy setting and project setting for vanilla PAL 25 fps SD stuff. All of the audio was correctly brought in -- then I disconnected all of the clips, made them dual mono, reconnected them, and copied them over to my old gristly project folder.
Now I can marry these dual mono 25fps files to the video files without having FCP incorrectly compensate for "30fps" by speeding these clips up slightly, so things stay in sync.

Similar Messages

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

Variable bit rate audio assets cannot be imported

Extracted some ac3 files from personal content DVDs I authored and burned myself (using DVDSP) to make a compilation. When I try to import these 5.1 ac3 assets, DVDSP is refusing stating: "Variable bit rate audio assets cannot be imported." These ac3 files play fine in QuickTime and other programs that handle ac3 files.
Can anyone help?

Hi:
Please give us some info about the original movie/audio formats and what encoding presets you used in Compressor.
Alberto

CP4 unable to decode and import the selected wav/mp3 file

I am trying to import a .wav file into a project and get an error message 'unable to decode and import the selected wav/mp3 file'.
I have tried the solution huan_miguel posted a week ago for someone with the same problem.
http://blogs.adobe.com/captivate/2009/03/adobe_captivate_4_installatio n.html
executed all the steps described in this link.
Unfortunately, the problem is still there.
Any other solution?

You're probably going to need to do both.
Although Captivate can certainly accept a 10 meg audio file, you would need to add it to the movie and then synchronise the slides to the audio. Chopping the audio file up and adding the relevant bits to each slide has both advantages and disadvantages. It's easier to control the playback synchronisation when you divide up the audio, but it's more work.
Take a look at this graphic from SoundBooth when I export an audio file recorded in Captivate 5:
Notice that it shows the bitrate of the Captivate WAV file as being 16 bit. This is quite a low bitrate. You can go as high as 32 bit in SoundBooth, however, in my experience, when you try to import a WAV into Captivate that is over 16 bit you will encounter issues. Cp just doesn't seem to like it.
If you want to learn more about bit rates in multimedia: http://en.wikipedia.org/wiki/Bit_rate
Suffice to say, you need to find out what the current bit rate of the audio file is, and if necessary, save it as 16 bit for Captivate. For this, you will require some kind of audio editing software.
You can try using Audacity as someone else suggested: http://audacity.sourceforge.net/ and try exporting the file out as 16 bit.
However, if you are going to be doing quite a bit of this type of work, I would suggest you invest in Adobe Soundbooth, or something of that ilk. You'll enjoy the experience more.

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?

Error in Adobe Captivate 4: "unable to decode and import the selected wav - mp3 file"

Good day!
I have a problem. When i want to import music file from standart Gallery, i see the mistake: unable to decode and import the selected wav - mp3 file.
I can't import mp3,wav files...
In Adobe Help site i have found this: http://help.adobe.com/en_US/Captivate/4.0/Using/WS5b3ccc516d4fbf351e63e3d119e9582190-7fd8. html
But in my Adobe Captivate 4 directory i don't have file regsvr32 NSAudio.dll.
Question: where i can download this file?
Thanks!

Hi there
As I recall there were two different "fixes" for audio with Captivate 4. One involved replacing the dll and another had something to do with installation problems on Windows XP. So give the link below a try.
Click here to view
Cheers... Rick
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Unable to decode and import the select wav/mp3 file

Hi,
I've been TRYING to test the Text-to-speech feature in
Captivate 4. Problem: "Unable to decode and import the select
wav/mp3 file" each and every time I want to use it. Frustrating!
I also tried importing sounds from the library, same problem.
My system is Vista Business 64Bit, 4Gb RAM, Intel SSD hard
drive, conexant high definition audio 221 on a Lenovo T400.
How do I fit this? I want to try it out damnt :)

I went through the "unable to decode and import the selected wav/mp3 file" error. After talking with an Adobe Rep, I quickly realized that I could NOT use the directions. I have no experience in using Command Prompts. I have written down the directions by key stroke for those people that are similar to me. Enjoy!
For Vista and Win7,
Click on the circle start menu,
In the search box, type in command and in the generated list, "right-click" on the comman link and change to "run as administrator".
Go back to the search box, type in command and hit enter. This will take you to the command prompt window (black window w/ white text) lol
You have to get to the right directory first.
Type: cd (space-bar) .. (hit enter)
Type: cd (space-bar) .. (hit enter)
If Captivate is installed on Program Files (x86) ---
Type: cd (space-bar) progra~2 (hit enter)
If Captivate is installed on Program Files
Type: cd (space-bar) progra~1 (hit enter)
Type: cd (space-bar) Adobe (hit enter)
Type: cd (space-bar) Adobe (space-bar) Captivate (space-bar) 4 (hit enter)
Type: regsvr32 (space-bar) NSAudio.dll (hit enter)
If you get an error at any point, re-key that particular line.
Once you see that the last line of command went through, open up Captivate and test the Text to Speech. These instructions fixed both computers I was working on!!!!

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!

Query re sampling rates when importing from CD in UK

Hi again. I was going to try to import a track from a CD via iTunes and convert it. I know that the rate is 44.1 before converting it in iTunes, but i'm unsure which raye to con vert to. my FCE setup is Pal 32Khz, therefor should I be converting to that rate to import into my project?
Again your help is much appreciated. Denis

Thanks Tom. I was perusing the manual while I was waiting read the bit about checking the Seqence settings. on going to Sequence>settings I noted that the audio rate was 32.0 KHz and the format was 32bit floating point, which it states I should match.
Thank again from one of the newbies following you Editing workshop and your video 'Final Cut Express HD Made Easy'

Sync and frame rate problems importing video to Flash CS3

Just like the subject states, I am indeed having frame rate problems importing video to Flash CS3.
When I export a video from Premiere with the settings:
1280x720
frame rate of 23.976
using H264 or Animation (basically every codec produces the same results),
"embed video on SWF and play in timeline"
When I do this the file exports fine. Playing it in Quicktime works fine and the audio syncs up no problem. But when I import it into Flash using the same settings, for some reason the movie clip is now too fast in that the video does not match the audio and seems to be at more like 12 fps now (just a guestimate). The action of the video is over long before the audio even comes in.I have verified that the frame rate is correct in Flash so it's not that.
I've done this literally hundreds of times in the past and suddenly it does not work and I can not figure out why. I'm not even really using video, but actually artwork that I am putting together as an animatic for an animated piece so the video could not have been imported incorrectly or digitized wrong either. Can someone out there please tell me how to make this work? Any help would be GREATLY appreciated!
-Mike

Thanks for you quick response however it really doesn't apply to what I am doing in any way. I will not be uploading the video to a server. I am using the imbed function so I can use it as a layer and then animate over it. It is just reference and will ultimately be deleted. The Fla file does indeed carry the same specs as I stated in my first post. I have done it this way dozens of times and I am trying to figure out why it suddenly will not work.
~Mike

External Audio recording to be imported & used in FCE

Hi there,
I am thinking of using this audio interface to record some vocals. It is a Focusrite Saffire 6 USB. It comes with a software called Ableton Live that can record the sound in wav or aiff format.
http://www.focusrite.com/products/audiointerfaces/saffire_6usb
According to this spec, it uses a *24bit 96kHz a/d covertor*.
http://www.htfr.com/more-info/MR326826
+TECHNICAL SPECIFICATIONS:+
+2 channel USB audio & MIDI recording interface+
*+24bit 96kHz a/d & d/a converters+*
I am using a standard defintion miniDV tape video camera and in my FCE, in the Easy setup, I have selected DV-PAL 32kHZ Amorphic.
My question is, will the sound recorded using this Focusrite Saffire audio interface be imported into FCE4 for editing together to mix together with my video clips (recorded in my standard defintion video camera) ?
Thanks you.
Cheers

RyanManUtd wrote:
QUESTION 1:
I also discovered something strange when I check the settings of all the previous video & audio clips imported into FCE4. Those that imported in from video camera, the values are 16 bit 32 kHz (How come ? My video camera is set to 12 bit 32 kHz)
Despite the material is recorded in 32 bit, the capture process with 48kHz setting are allowed, but on long shot, you can expect a audio drift from video if sequence's audio is not the same as original sources.
QUESTION 2:
Then, I look into all the audio clips I bought from iTunes & imported into FCE4, they are all listed as 16 bit 48 kHz.
The funny thing is that both these type of different clips (ie, 16bit/32kHz & 16bit/48kHz) can be rendered without problem. How come ? FCE4 can mix clips of different audio types ?
BTW, importing iTune purchased song into FCE is prohibited by iTune EUA. Any non-natively supported media can be rendered in FCE, but as explained in question 1, you can expect audio drift on long clip. This problem is not relevant with music track audio which is it not part of a video footage.
QUESTION 3:
If I set the Ableton software output to 16bit 48 kHz, does that mean that I actually cannot make use of the hardware abilities to produce sound of 24 bit 96 kHz ? Is there much difference ?
96 kHz is used for professionnal multi-track audio editing. You have to use high-end audio monitors to ear the difference... if you can. Usually only audio engineers can ear the difference.
Audio CD use 44 kHz and the quality is pristine on hi-fi deck. 48 kHz is the standard in video production and even used in dolby 5.1 home theater configuration.
QUESTION 4:
In the Easy Setup, I cannot find 16 bit 48 kHz. I only have the following option in the Easy Setup of FCE4:
a) DV-PAL
b) DV-PAL 32 kHz
c) DV-PAL 32 kHz Anamorphic
d) DV-PAL 32 kHz Anamorphic Basic
e) DV-PAL 32 kHz FireWire Basic
f) DV-PAL Anamporphic
g) DV-PAL Anamorphic Basic
h) DV-PAL DV Convertor
i) DV-PAL FireWire Basic
So, which one do I chose ?
Use DV-PAL.
Michel Boissonneault

Audio out of sync importing from panasonic PV-GS500

i'm importing video from a panasonic PV-GS500. each time i try i get an error message which 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 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."
sometimes this is true (it's totally out of sync from the start) and sometimes it seems ok. but i don't know how to figure out the audio sample rate of the tape. so how do i match it with the capture preset?

Hi. This is the exact problem I've been having trying to capture footage from my PV-GS500. Did you ever find a solution? Or does anyone have a solution to a problem like this? I'm over my head with this problem. Thank you.

Convert Sample Rate on Import- does not work

Setting: *"Automatically convert sample rate of imported audio"*
You'll want to set this if you have audio on your computer
that wasn't sampled at 44k. It prevents your imported audio from
sounding like Mickey Mouse drinking a gallon of espresso and inhaling helium.
HOWEVER...It does NOT work when you drag audio into the project from
Logic's internal File Browser.
Dragging audio into the project from Logic's Browser,
or browsing to the file in Logic's Browser and choosing to
"Add selected Files to Audio Bin",
will NOT convert to the audio's sample rate.
Sample rate of your audio is only converted if you do: "Import audio file",
then browse to the file(again) in a popup browser dialog, and then choose import.
If the audio is already in your Audio Bin, choosing "Convert File"
will bring up another Finder Dialog asking you where you want the
converted audio to be placed.
Now you'll have to Browse (yet again) to the song's folder, because that's where
you'll probably want it anyway, and choose to put
your converted audio into the song's folder.
Please realize: If a composer or producer wants his/her audio to sound like
Mickey Mouse on Helium and Espresso, he will do it after the fact, on purpose, using a plugin.
Is it too much to ask, to have this Setting do what it says it does, with ALL audio brought into the song?
No matter how I bring audio material into a Project, the sample rate should always conform
to the Project's sample rate.

Hi there
I've been using Captivate since version 1. And to be honest, I'm not sure I've ever seen the option work. If it does work, I'd also like to know under what conditions it works. Perhaps it only works with certain applications. I'm not sure. Largely I've ignored the option exists because I've never recorded anything where I wanted tooltips to be an option. I've never wanted to simulate to that detail.
You should probably consider filing a bug report.
Cheers... Rick
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Capturing-audio sample rate

I have fce hd v 3.5. My camera is a sony dcr-trv11. I am trying to "capture now" but it looks like I am dropping frames. When i hit "esc" it stops capture of course, but I get a message that reads "the audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video ans audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape." Okay, so how do I do that? When I originally captured my footage with imovie I had no problems. I had trouble importing them to FCE so I scrapped them all and decided to start from scratch with FCE. I have 199.5 GB of free space.

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

Geting audio sample rate error, help

Hey all, been doing a massive project where I ma bring in tons of old 8 mm tapes, hi8 and digital 8. This one tape I brought on though however is giving me grief. I keep getting this error:
The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape.
If find this odd though cause the tape should be standard, NTSC dv 48 K. Any suggestions. Also how do I reset my final cut so that when I plug in a camera it always reads it as what it recognizes. I ask this because I had to used the setting uncontrollable device because on the original old 8mm tapes there is no time code so I had to capture that way.
Anyhow, any suggestions on this would be great for if I recapture. Cause I could line it up by eye but want to find the problem so I know for the future. Thank you.
Nathan

This is a recent problem that seems related to a recent upgrade of QuickTime. Here's why.
In the last month, a rash of these posts have begun to appear:
"DV Capture Audio problem"
http://discussions.apple.com/thread.jspa?messageID=6708693&#6708693
"audio/video"
http://discussions.apple.com/thread.jspa?messageID=6591262&#6591262
Plus this thread, plus my own.
In my case, nothing changed in my operating system or Final Cut Pro version. I upgraded to QuickTime 7.1.6, and the problem began. I have upgraded all the way to 7.4 to no avail. When I attempt to import a DV clip using the same Sony DVCAM deck that imported the same clip in December, I get the error every time. Nothing has changed in the tape, the deck, the Project or Final Cut. I am simply unable to import DV video. I can import other kinds (Panasonic P2, for example), but DV is a no go. I cannot get rid of this error.

Audio TC Rate woes -- importing no-TC wav as 25fps instead of 30fps

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