How to adjust sample rate in "capture preset"?

I'm having to recapture footage shot some years ago on my old Sony VX1000 using my Sony Z5U. I'm working on FCE 4.0 (OS 10.4), Easy Setup at DV-NTSC and capturing by "capture now." I'm getting a warning saying that my sample rate of capture does not match sample rate of source tape and advises me to ensure that my audio sample rate of the "capture preset" matches the sample rate of [my] source tape. Browser tells me that my sequence and my clips audio rate are both at 48 KHz, but my sequence audio format is 32 bit floating point, and my clips are at 16 bit integer. How do I access capture preset and synch capture rate?
Thanks

48KHz = 16bit
32KHz = 12bit
Can't be both at the same time.
Most cameras come set to 12bit as the factory default. On the Canon miniDV cams like your ZR500, go into the menus for *Camera Setup > Audio Setup > Audio Mode* and you should be able to toggle between 16 bit and 12 bit audio.
You want to film your videos using 16bit audio if at all possible. If you have done so, then you should use the DV-NTSC easy setup in Final Cut Express (not the DV-NTSC 32KHz easy setup).
If you inadvertently filmed your video using 12-bit audio then you will need to use the DV-NTSC 32KHz easy setup in FCE.

Similar Messages

Matching sample rates of capture preset and tape

After upgrading to FCE 3.5, I don't have the dropped frames problem anymore and the audio is in sync but every time I'm done capturing a clip from my camera I get an alert saying the audio sample rate of captured media files don't match the sample rate on source tape.
So, I gather that the sample rate capture preset must equal the sample rate of the tape. So I need to know two things:
1. How do I set the sample rate capture preset? I've looked everywhere I can think of.
2. How do I find out what the sample rate of my tape is? I've looked in my Sony FX1's settings and can't find anything.

Hi(Bonjour)!
Your FX1 sound settings can be found in audio menu, select 16 bit if you shoot in DV format.
Audio settings for HDV is not adjustable.
Your sequence and capture settings have to match the properties of your captured clips.
Select DV-NTSC 48kHz in easy setup. (for DV)
or Apple intermediate codec 1080i60 for HDV.
Michel Boissonneault

How to adjust sample rate of data?

I have some data collected at 1683 Hz (yes, that was what I had!) and would like to reduce the sampling rate to some meaningful number, say 1024, 500, 400, or similar.
What should I do?

Well, the calculation is the approximation of your channel (variables with index 0) to a new one (index 1).
The freq(0) and freq(1) are the sampling frequencies for the channels for the case you have waveform channels.
The n(0) and n(1) are the numbers of the data points inside the channels. The new created channel should have the number n(1), calculated from n(0) with regard to different sampling ratios.
The real code is the line Call ChnSplineXYCalc(..... The properties swapping can be commented out, but then the new channel would have a "system name", something like "Approximated XY", and the same for description and units... Probably one can avoid it by changing of settings, but I use to do it by code.
In short, here you copy the properties from the "old" channel and paste them to the new one.

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

Adjusting Sample Rates

I just got Audition CC 2014 and when I try to record it says the sample rates of the input and output devices don't match. How do I adjust those rates in Windows 8.1?

Okay. You need to go to the Windows Sound control panel and select the Playback tab. Highlight your output device (if you have no external card there may well only be the one) and select Properties. Then click the Advanced tab and there'll be a drop down menu for bit depth and sample rate. I'd suggest 16 bit/44100Hz (CD Quality) for pure audio that will end up on CD or MP3 or 16 bit/48000Hz if it's going to end up on video.
Next, go back to the Windows Sound control panel, click the Recording tab, and follow the same procedure (highlight, Properties, Advanced) and select the same settings in the drop down menu there,
This should sort you out. In the longer term, consider an external USB audio interface--it'll be hugely better quality and, if it has ASIO drivers, it'll stop Windows applications from changing audio settings without telling you.

How to improve sampling rates in mega samples spartan 3e board

I am trying to acquire data in spartan 3e board in mega samples rate can you post me any examples for this purpose
Solved!
Go to Solution.

If you're trying to get analog samples in that range, the issue that you're going to run into is that that sort of sampling rate is right at the bleeding edge of what you're asking for (it's a 2-channel @ 1.5Msps chip, the LTC1407A-1). If you actually want to capture at these rates, you're going to need to pull some interesting LV-foo using single-cycle timed loops in the FPGA diagram or drop some code down into an IPNode/CLIP Node, with outputs from those IP blocks hooked to the IO in a SCTL.
Please let us know a bit more about your specific design.
-Brad

Adjusting Sampling Rate to write continuous data to excel?

Dear colleagues,
I am reading data from a DAQ from 4 channels and am wanting to write the data continuously to an excel sheet. Problem is, I cannot get the sampling rate and # of samples to appear "constant". instead, labview is only sampling x number of samples in a row and storing those to excel. What I want is this:
Beginning at time t=0, and every 0.2 seconds after that (5 Hz), read the values from all 4 channels and append them to an excel file. What I am using is a 1D to 2D array converter and the Write-to-Spreadsheet function outside my while loop. If I set it to only do 1 sample every 0.2 seconds, then it just stays at t=0, just refreshing the value and thus overwriting my excel data. How can I make it sample continuously from t=0 to t=when STOP button is pressed, and then log all samples into excel? I have attached my sample vi below. Thank you.
Attachments:
working_final.vi ‏547 KB

Hi Mike, Thanks for replying.
The latter of the two methods you mentioned, " you write your values every iteration to the file, therefore you have to place the "Write vi" inside the while loop.", this would be ideal. What sort of conversions need to be done in the while loop before the data is written? In my file, you can see that I am trying to use a converter to go from Dynamic Data to a 1D array. Would this be the appropriate method for doing this? What I am trying to accomplish can't possibly be this complex, I'm sure lots of people require the use of a similar write-to-spreadsheet function.

Problems with audio sample rate when capturing DV50

Hello from Spain, I have a problem when capturing DV50 source footage: I setup my sequence settings and capturing settings like always, DV50 codec and 48k sample rate for audio, then when the capturing process finish the result is a file that contains audio to a non-48k audio sample rate, then video and audio don´t match. The sample rate is 47945 and other similars, not always the same, sometimes 47214 sometimes 48000, and there is no rules...My capturing device works ok in other computers and the footage has been capture ok in other Mac...I don´t know what happens.
Sebas, kind regards
G5 Quad Mac OS X (10.4.6)

Hello!, first of all thanks for your reply. Yes, the camera is the same, the problem is on my computer, I have discovered a solution: first I capture the audio, only the audio, and then the video, only the video. After this I soncronize the two separate files on Final Cut timeline. It´s a slow process, but I am really desesperated. I don´t know what happens, I few minutes ago I have captured a 30 mins. video and the sample rate of the audio was 42.145k (???). It´s a crazy thing...

How To Lower Sample Rate

Is it possible to adjust the sample rate lower than the default when attempting to record an audio source?

44.1 is the lowest sample rate to record audio at in Logic. But put a Bit crusher plugin on the track and change the resolution there for more lo-fi sounds.
JG

Sample rate of capture audio

I found I can change sampe rate capture audio into 8000Hz by using dsound://8000. But the sound is still 16bit and stereo.
Anyone know how to change it into 8bit and mono?

I think you are right. Jmf has to play with those dll files in windows. But I have already installed the performance packs. I just got the information in jmf log which says the soundtrack can't change because it has been specified to RAW type. So I think it probobly due to the way jmf play with capture audio device. Once the audio format has been defined to capture audio device.(dsound://) It can't be changed.
I am developing a audio one-to-one chat applet which is suppose to run on windows at this stage. I will consider cross platform solution later on.
Anyway, I found an alternation which is using dsound://8000 and dvi/rtp. It can generate a 8000Hz, 4BIT MONO sound for me. It fairly good. However, I am still thinking about how to cut it down. Might be 4000Hz.... Anyone konw what it sounds like? Is it still acceptable for voice transfering?

BT-183 bluetooth: how to adjust data rate?

Hello everybody.
The title of this topic is pretty self-explanatory.
Is there a way to increase data transfer speed of ASUStek BT-183 bluetooth module?
28 KB/s, which is set by default, is not sufficient for me
Even my mobile phone can transmit data twice as fast.

Well, the calculation is the approximation of your channel (variables with index 0) to a new one (index 1).
The freq(0) and freq(1) are the sampling frequencies for the channels for the case you have waveform channels.
The n(0) and n(1) are the numbers of the data points inside the channels. The new created channel should have the number n(1), calculated from n(0) with regard to different sampling ratios.
The real code is the line Call ChnSplineXYCalc(..... The properties swapping can be commented out, but then the new channel would have a "system name", something like "Approximated XY", and the same for description and units... Probably one can avoid it by changing of settings, but I use to do it by code.
In short, here you copy the properties from the "old" channel and paste them to the new one.

FCE Capture-sample rate error

After capturing Digital8 footage off my Sony DCR-TRV310, I get 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 media files to be out of sync. Make sure the audio sample rate of capture preset matches the sample rate of your tape."
So far, in spot checking the media it seems to be OK, but I'm anal about errors! I couldn't find the audio sample rate for the camera/tape in the sony manual, any ideas? Also were do I select the audio sample rate of capture preset, even if I knew the tapes sample rate? Or is this nothing to worry about
Thanks, Schooner

Hi(Bonjour)!
Usually, camcorders allows to set the audio sample rate to 12 bit (32 kHz) or 16bit (48 kHz).
The audio sample rate for DCR-TRV 310 can be set to 12 or 16 bit. See page 57 in this user manual:
http://www.mediacollege.com/equipment/manual/sony/camera/dcrtrv315.pdf
FCE easy setup allows to select DV-NTSC 32 or 48 kHz for sequence. This settings apply to capture session too. The message you've got means that the audio sample rate on tape doesn't match the sequence audio format.
This problem may give a drift synch between audio and video.
First check the item format for video clip in browser. The audio sample rate should be 32 or 48 kHz. Note it and select the right setup in easy setup dialog. Create a new sequence and insert your video clip.
Michel Boissonneault

How to know the sampling rate for NI6624?

Dears,
I am trying to measure a transient signal that is a time-vary counter train. The target frequency is increased from 0 Hz to 50 Hz when the measurement time rises to 1 s from 0 s. The NI 6624 card and the LabVIEW DAQmx have been adopted. In the block diagram, the terminal of measurement method “Low Frequency with 1 Counter” is set in “DAQmx Create Channel (CI-Frequency).vi”, and the “Finite Samples” mode is chosen in “DAQmx Timing (Implicit).vi”. Then the transient signal points (increasing-frequency points) will be got successfully within 1 s. Now I have a question: how do I estimate the time step “dt” between these data points? Knowing the default sampling rate of the card seems a better way to help me to define the "dt", and calculates the time stamp at each data point. If the foregoing concept is true, how the internal sampling rate in NI6624 obtains? Beside, for the transient counter signal, any way to get the time stamp of data points is also welcome.
Thanks for anyone comment,
Adan

Adan,
When selecting "Implicit" as the DAQmx Timing type, you are indicating
that a data point will be taken for every measurement the counter
performs. When you create a task of type "Low Frequency with 1
Counter," the counter simply uses the card's internal timebase to
measure the period between edges of your signal. It then takes this
period measurement and converts it to a frequency. Therefore, the
spacing between the samples you read out is simply the inverse of the
subsequent frequency measurement sample.
Hope this helps,
Ryan Verret
Product Marketing Engineer
Signal Generators
National Instruments

Drop the sample rate

I'm trying to drop the hardware sample rate. It's at 44800Hz. When I want to capture audio through my Audigy 2 soundcard at 44100Hz I get the message "We do not support recording when your file does noyt match your hardware sample rate. Your current hardware sample rate is: 48000Hz."
Edit -> Adjust Sample Rate does nothing.
Edit -> Hardware Audio Setup shows a 48000Hz sample rate but I can't change it.
Oh. Is there a "global" reset on settings?
Thanks
Gis

>I'm trying to drop the hardware sample rate. It's at 44800Hz. When I want to capture audio through my
Audigy 2 soundcard at 44100Hz...
My very strong advice to you is not even to try. Audigy soundcards work internally at 48k, regardless of what you tell them - the internal architecture is fixed. If you tell them to work at 44.1k, then all they do is 'spoof' the output by doing an internal sample rate conversion on the fly. The problem with this is that it's not a very good conversion - and Audition can do it rather better. So the way to go is to record at 48k, and then do the conversion rather more transparently in Audition. Doesn't take long, and won't degrade the sound even more than the card already has. So...
...a better way to go would be to ditch that pile of Creative rubbish and get a proper soundcard!

Capture dvi and optical digital audio-in through one custom capture preset

How can I create a FCP capture preset that will capture from a digital video to firewire box (grass valley (formerly canopus) advc-hd50) AND the optical audio in port on my desktop mac dual g5?
I have HD dvi video (which has no audio content) and digital audio (or analog for that matter) that i wish to capture in sync then edit in FCP.

if you go to audio/video settings you can duplicate a setup that is close to what you want and edit that setup. under capture presets> audio select whichever input on the mac you wish to use.
consult your documentation with the capture device for the video side. when you have something that works save that as an easy setup (on the summary tab "create easy setup")
I've never used either the optical nor analog in so I can't vouch for it's behavior however it is available as an option
Hopefully your device will somehow pass timecode so edits are repeatable

How to adjust sample rate in "capture preset"?

Similar Messages

Maybe you are looking for