Multiple sample rates!!!

I am doing something I have done a million times but this time I am unable to do so! Here it is.
Pull a clip on to the timeline viewer, extract the audio, split the video clip in half, trash one of the halves, insert other clips to fill in the video, turn the audio down to 0% on the new clips. When I try to export I get a failure that says I am trying to export multiple audio sample rates. I am doing nothing different than earlier today, but now it won't let me. I have tried several variations of this work to fool the computer but nothing works. I am moving on to another movie but if I run into this problem again it's off to Final Cut Express with no regrets.
Any ideas?
I thank you in advance.
Steve

I couldn't reproduce your error although I tried to follow your workflow (I had to use PAL, though).
Anyway, there is an audio sample rate bug in iMovie 6 because it extracts audio at 44.100 kHz instead of 48.000 kHz:
DV works best with 48.000 kHz, 16-bit audio (32.000 kHz, 12-bit audio can produce problems). If the DV audio is 48.000 kHz (as it should), and you use the Advanced/Extract Audio -command, the extracted audio is 44.100 kHz. (You can find the extracted *.aiff audio files at iMovie project package's /Media -folder).
http://www.sjoki.uta.fi/~shmhav/iMovieHD_6_bugs.html#extractaudio
FWIW, iMovie 4 also had an audio sample rate bug:
Imported audio in .mov and MooV files is converted to 32 kHz:
http://www.sjoki.uta.fi/~shmhav/iMovie4bugs.html#32khz

Similar Messages

Multiple sample rate question

I have one project with edits of different files all over the place recorded at 44.1 kHz. I have another project with edits all over the place recorded at 88.2 kHz. I wish to combine all tracks from these two projects. I realize I will need to convert the sample rate of all the 88.2 kHz files, but I don't want to mix down any track. I want all of the edits to be cut at the same places in time, just with the referenced files being at a different sample rate.
I've tried everything. It seems that it is impossible. If I convert the sample rate of the referenced files, then open the project and change it's sample rate, the edits are now cut at differnt locations in time. It seems I have to mix down each 88.2 track, then sample rate convert these to 44.1.
But I need to have all the edited segments as they appear in both projects in the same project. Any ideas?

Don't record at 88.2! Why would you do that?!
I'd back up and archive your current session first as you are probably going to have to do some very destructive messing.
I haven't put this to the test but maybe you could try converting the route files in a third party software like ProTools. You'll get a bunch of prompts when you reopen Logic but this may work. But for me the simplest (albeit time consuming route) would be to convert all of your regions to new original sound files (you can do this from the audio pop down on the arrange page). Then in your audio pool select all unused audio, delete and go through each new file and convert the sample rate. It's a bit of a car crash situation needless to say next time preparation preparation preparation!
Finally when you have been through this kind of process make sure that you are operating at the correct new clock rate before continuing otherwise this stuff still aint gonna work.
Hope this is of help.
Good luck.

Save multiple sample rate data to TDM file

Hello, LV connoisseurs
I use 2 Multifunction boards and LV 7.1 to gather slow and fast data simultaneously, 'slow' being 10 temperatures at 150 Hz, 'fast' being 12 pressures at 15 kHz, rate factor between slow and fast is constant 100. My acquisition is set to 'continuous' with blocks of 15 and 1500 resp., so each sample set takes 0.1 sec.
Currently, I use 2 loops, one for each board. Slow data are written to .lvm, fast to .tdm, and this works fine.
But I wonder if in this configuration it might be possible to
- use one loop only (yes, trivial) with the main target being to
- write data into two channel groups of one .tdm-file, one for the 'fast' the other for the 'slow' data?
If at all possible, would this require the consumer/supplier scheme such as to allow the interspersing of data or can I do this directly?
Thank You for your input.
Michael

Just to make the distinction, you should probably be using TDMS (the S stands for streaming) instead of TDM if you are continuously writing data. TDM is more for writing a snap-shot and doesn't work so well for continuous data (big memory leaks last time I used it back in '06).
Also, as no time data is stored you probably also want separate timestamp channels then for your fast and slow data.
As mentioned, it is no problem having multiple writing loops using the same TDMS reference.
nrp
CLA

Error message stops .wmv creation: "Inconsistent Audio Sample Rate..."

Just recently when attempting to export a video into the .wmv format, QuickTime Pro 7.2 gives me this alert message, "Inconsistent Audio Sample Rate. The media you are exporting contains audio with multiple sample rates." QuickTime will not compress (export) to .wmv. It will export to .mov. and .m4v.
This problem is new. For more than a year my weekly video exported to the .wmv format without fail.
I am working on a Mac G4 Dual Processor and iMovieHD.
Any help you can offer would be greatly appreciated.
Mac G4 Dual Processor Mac OS X (10.4.10)

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

Cannot export as WMV - Inconsistent Audio Sample Rate error

Hello,
I am getting an error message when I try to export a sequence using quicktime conversion to a WMV. "Inconsistent Audio Sample Rate - The media you are exporting contains audio with multiple sample rates".
I used Soundtrack Pro for the first time- to analysis and fix an audio file with pops. It worked wonderfully. I don't have much experience with this program and the audio is set to 32-bit stereo with a file ext stp. Not sure how to change this or if I need to.
I have done some tests and I have no problem exporting as a quicktime. I am using the latest version of flip4mac software. Your help is appreciated.
Cheers,
Chris

Hi Susan,
Thanks for your quick reply. The link helped alot.
For others facing this problem, I found this solution.
1. Export sequence as a Quicktime movie (not conversion) using current settings and make movie self-contained.
2. Create a new sequence and change the sequence audio setting (under sequence-general) to 44.1 kHz - Dept 16-bit config default
3. Drag your Quicktime movie into this new sequence.
4. Export using Quicktime conversion to WMV with these settings for audio.
5. WMA9 Standard - One pass, CBR at 64kbps, 44.1 khz, Stereo.
Voila.
I have struggled with flip4mac from the start. Never had any luck using the 2 pass method which 1'm told is better quality. Tech support at the flip4mac discussion group generated much discussion but no solution. I have settled with single pass quality in order to get the job done.
Does any remember the days of linear editing when you delivered a master, protection copy and some VHS dubs and you were done? Oh, I'm showing my age. Ha.Ha.
Cheers,
Chris

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 does the sampling rate for my DAQCard-700 max out at 6578Hz regardless of the number of channels I'm sampling?

My DAQCard-700 is acting funny. I am using LABView 6.0.2 on a Dell Inspiron 8100 laptop for continuous analog input. For some reason my DAQCard-700 won't let me acquire multiple channels unless I create virtual channels in max and then run AI Config with a list of the virtual channel names. That's fine (but weird), but I max out my sampling rate at 6578 Hz, which is the max sampling rate if I would have sampled all 16 (Referenced, single-ended) AI channels. But I'm not -- I'm sampling 6 channels, as proven by the "number of channels" indicator in AI config and just looking at my data. If I sample just one channel, I can get up to 105248Hz. Is my card f
ried, or am I missing a hidden setting somewhere? I have the latest NIDAQ.

Hello;
By the description of your problem, everything is pointing us to a hardware failure.
There aren't any tricks to acquire multiple channels with that card.
The best way to go about this is to have NI to fix the board for you.
Regards
Filipe

How to build a array with high sampling rates 1K

Hi All:
Now I am trying to develop a project with CRio.
But I am not sure how to build a array with high sampling rates signal, like >1K. (Sigle-point data)
Before, I would like to use "Build Arrary" and "Shift Register" to build a arrary, but I found it is not working for high sampling rates.
Is there anyother good way to build a data arrary for high sampling rates??
Thanks
Attachments:
Building_Array_high_rates.JPG ‏120 KB

Can't give a sample of the FPGA right now but here is a sample bit of RT code I recently used. I am acquiring data at 51,200 samples every second. I put the data in a FIFO on the FPGA side, then I read from that FIFO on the RT side and insert the data into a pre-initialized array using "Replace Array subset" NOT "Insert into array". I keep a count of the data I have read/inserted, and once I am at 51,200 samples, I know I have 1 full second of data. At this point, I add it to a queue which sends it to another loop to be processed. Also, I don't use the new index terminal in my subVI because I know I am always adding 6400 elements so I can just multiply my counter by 6400, but if you use the method described further down below , you will want to use the "new index" to return a value because you may not always read the same number of elements using that method.
The reason I use a timeout of 0 and a wait until next ms multiple is because if you use a timeout wired to the FIFO read node, it spins a loop in the background that polls for data, which rails your processor. Depending on what type of acquisition you are doing, you can also use the method of reading 0 elements, then using the "elements remaining" variable, to wire up another node as is shown below. This was not an option for me because of my programs architecture and needing chunks of 1 second data. Had I used this method it would have overcomplicated things if I read more elements then I had available in my 51,200 buffer.
Let me knwo if you have more qeustions
CLA, LabVIEW Versions 2010-2013
Attachments:
RT.PNG ‏36 KB
FIFO read.PNG ‏4 KB

Low sample-rate measurements on the PCI-6115 DAQ card

I need to measure an analog signal at a sampling rate of a few tens to hundreds of Hz in sync with the rising edge of an external clock. I have a PCI-6115 DAQ card w/ Labview 6.1 and NI-DAQ 6.9.2. The PCI-6115 is a high speed card and has a minimum sample rate of 10 KS/s. Is there any way of implementing a low sampling rate measurement using the PCI-6115 in sync with an external clock?
Thanks in advance.

Kuldeep,
It is possible to do what you are describing above (in fact I don't think an external clock is required to do this), however, bear in mind that the reason for this minimum sampling frequency is due to the ADCs used on this high speed board. The ADCs used are pipelined ADCs, meaning that when a signal is digitized, it is digitized in distinct stages within the ADC (in the case of the 6115, I think there are 3 stages involved). Data is moved from one stage of the ADC to the next each time a sample clock pulse is recieved. If too much time elapses between these clock edges, the signal to be digitized can actually 'leak' off of the ADC. This can result in improper digitization, which can lead to less accurate measurements. So, while it is possible to mak
e the device sample below it's minimum rate, it may be advisable to sample faster than the rates required by your application, and either average multiple data points per measurement, or throw away extra points taken.
I hope this helps,
Dan

Different Sampling rates for different channels in Analog Input

Hi,
I would like to acquire data at different sampling rates on different channels say ACH 0, ACH 1 and so on. I have a PCI 6052E board and NI DAQ 6.9.2. Also is it possible to simultaneously perform Analog output on two different channels along with the Analog input? What will be the problems/consequences as far as the system resources are concerned. I am a beginner in this area and would greatly appreciate any help/pointers for my queries.
Thanking you in advance
Deepak

Search the eaxamples that ship with LV.
Theer is one called simultaneous input and output or something like that.
It will get you started.
re: multiple scan rates. This is acoomplished by sampling all channels at the highest rate and throw away the expttra samples you do not need.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

Can i use double buffering with SCAN_Sequence_Setup and sample rate divisors?

If possible I would like to use double buffering when acquiring multiple channels at different rates using SCAN_Sequence_Setup. What are the tricks to sizing the buffers, if any.
PCI-MIO-16Xe-10 and PCI-6052E are the boards I'm writing for, using Borland C++ builder under Windows2000
Thanks,
Brady

Hello;
At this time, you can't set up a multiple Scan Rate for different channels at same DAQ board. But, you can set up your Sample Rate to the fastest required to acquire you fastest channel, and then discard the readings of the other channels and only take the values for the different channels at their theoretical rate.
Hope this helps.
Filipe

Having trouble with wav files and sample rates

Hi ,I am having trouble with wav files and sample rates .I have been sent multiple projects on wav as the main instrumental ; I wish to record in 48.000kHz .Now comes the problem.When I try to change the project to 48k It seems to pitch up the track.I can't have them send the logic/project file as most have outboard synths,different plug ins etc.This particular case the producer has recorded the synth task in 41.000 kHz .My successful outcome would be to be able t create a project file in 48 kHz .And NOT pitch up whne I add the instrumenta wav file .Any help would be gratefully recieved,this is my first post so any mistakes I may have made go easy

You'll have to convert the actual synth audio file file that the producer gave you to 48kHz. You can do this in the audio Bin in Logic.

Audition 3 seeing a different sample rate setting than what the device shows

Hi,
I have just installed Adobe Audition 3, along with the 3.01 patch, on a brand new system running Windows 7 64 bit. The mother board is an Asus Sabertooth X58 using Realtek High Definition Audio. The device drivers show that the audio sampling rate for line input is set to 24 bit 192K. I wanted to set it to the maximum that the sound card would allow to test performance and audio quality.
The problem is when I bring up Audition 3 and hit record, I get the message "We do not support recording when your file does not match your hardware sample rate. Your current hardware sample rate is 44100Hz". Clearly this is not the case since the Line In Properties - Advanced tab is displaying "2 channel, 24 bit, 192000 Hz (Studio Quality).
Under Audition's Audio Hardware Setup it shows only one choice for Audio Driver: Audition 3.0 Windows Sound. It also displays Sample Rate: 44100Hz, Clock Source: Internal, Buffer Size: 2048 samples with no way to change these values.
If I click on the Control Panel button I get:
DirectSound Input Ports:
Device Name: Line In (High Definition Audio Device
Audio Channels: 2
Bits per Sample: 16
Anyone know of how I can change these settings to get Audition to agree with the device settings?
Thanks
Dale

DaleChamberlain wrote: Anyone know of how I can change these settings to get Audition to agree with the device settings?
I'm afraid that life is nowhere near that simple. The main issue here is that Audition, in common with most audio software, uses a driver system called ASIO to talk to the sound device - this cuts out a lot of the OS and reduces the latency of the system considerably. There are several problems with ASIO though - the first being that it only supports a single device per system (or sometimes multiple identical devices if the manufacturer can make them look like a single device), and with software designed to use this driver, then to use any other driver (like a native Windows one) you have to use a converter stage like ASIO4ALL. This will convert the ASIO streams to WDM, and let you use multiple sound devices - but with increased latency.
It's the second problem that's really going to stuff you though - and that is that quite reasonably, ASIO is limited by its inventors to run only at three sample rates; 44.1k, 48k and 96k. So there's no way you can run at what you think might be a higher quality setting. All settings above even 48k are making your sound device work much harder, and for what? All that happens is that you increase the potential frequency response to way beyond the human hearing range - to no purpose at all. You don't have sources that can produce useful output at these frequencies, and you certainly don't have the means to reproduce them. This has all been well documented and explained before, so I'm not going over all that again. In a nutshell, Nyquist points out that any digital sampling device has a frequency response limited to a maximum of half of the sample rate, so for 48k that gives us a frequency response up to 24kHz - comfortably higher than any adult can hear by quite a long way. Anything you sample and record beyond this by using even 96k is nothing but noise as far as humans are concerned, and unpercievable noise at that.
So what the line input properties tab is saying is, if you have a non-ASIO driver designed to support all potential rates, possible. You don't have an ASIO driver available, because it's a built-in sound device, and anyway you've already pointed out that it's using the Audition Windows driver (a cut-down version of ASIO4ALL, effectively), so a conversion is already taking place. What Realtek refer to as 'High Definition Audio' is no such thing - all on-board sound devices of this nature are of universally low quality, and to improve this you'd need an external device - of which there are many available, usually with dedicated ASIO drivers. But none of them will work with ASIO beyond 96k, simply because the standard doesn't support any higher rates.
If you download and use ASIO4ALL (it's free), then you will get an additional control panel which will show you exactly what your sound device is capable of doing as far as Audition or any other ASIO software is concerned, and this is a useful diagnostic tool anyway, so it's worth doing. You just select this option when installed, instead of the Audition Windows Driver.
I'm sorry to be the bearer of what seems like bad news, but actually, it isn't. You will percieve no quality difference at all running at anything beyond 48k sample rates; all you will be doing is wasting your computer's resources unnecessarily. You waste both processing resources and hard drive space by processing at ridiculously high sample rates, and there are zero returns.

Counter Problem at low sampling rate

What I am trying to do is to count the input from the camera and generate the number of pulses depending on the delay and interval. At the rate of 33ms the program works perfectly but if the sampling rate is 150ms it generates 2 pulses at a time. Is there a better solution to my problem ? Am I doing anything wrong ? Is it a problem with the loop timing or acquisition timing? I have attached the program. What puzzles me is that it doesnt work if the rate is lower! Please HELP.
Thanks
p.s. I am very new to LabView so please excuse my ignorance.
Attachments:
CoolSnap.vi ‏142 KB

Hello sha33,
Thank you for getting back to me with more information. After looking through your code once more, I believe I have found the issue. Since this application is timed entirely in software, it is based on how quickly your operating system can run the program, and thus cycle through loop iterations. Since you have no Timing VIs in the program, this loop is running as fast as possible. Now, in the 33 ms case, you are getting a TTL edge a lot faster, so the DAQ assistant is updating its count faster as well. Lets say hypothetically that your loop is iterating at 60 ms. This means that every time you run through the loop your DAQ assistant will read the updated count value, and since you are counting every 33 ms, your count value will have updated.
Now let's look at your 150 ms TTL signal. This is significantly longer than the 33 ms signal in the previous case. Lets say hypothetically your loop is running on average at 60 ms again. This time, your edge will come in and the counter will increment, generating a pulse with the second DAQ assistant accordingly. Now your loop will run through again 60 ms later but your next edge on the 150 ms signal is still around 80-90 ms away. This means that your DAQ assistant will return the same count value as the last iteration, and once again a pulse will be generated.
Obviously the numbers I used for the loop iteration timing are hypothetical and entirely dependent on the operating system and its resources (i.e. it will run slower with an MP3 or Movie playing in the background). My suggestion would be to use a timing VI such as Wait Until Next ms Multiple to clock your loops more consistently. The goal would be to make sure that your loop is repeating faster than your counter is updating. Alternatively, you could account for the possibility of no counter input in your overall application to prevent a false positive.
Let me know if I can clarify any of the above information, or if you have any additional questions related to this issue.
Regards,
Chris Behnke
Sr. RF Engineer
High Frequency Measurements

Sample rate/quicktime pro audio/video

Hi all:
Have a few questions related to Quicktime pro on mac related to sample rate of audio and video...any help understanding much appreciated...
I am a music composer using logic pro and recently started working with filmakers who send me quicktime files of the video to score to...
regarding sample rate of audio as played back in quicktime...I'm a bit confused....
When playing back either an MP3,wave,Aiff at 44.1k or 48k
seems that quicktime plays at the right speed and pitch of audio...Does Quicktime use it's own audio processor to convert sample rates in real time at playback?
When I play a video track with audio recorded at 44.1k or even an MP3...and playback on my soundcard set at 48k...it does play at correct pitch...even though theory of audio tells me that the sample rate is very inportant especially when playing back...sample rate needs to be the same on both the soundcard audio clock and the software...
Is this part of the magic of quicktime or am I missing something.
Even a bit more confused when I got a dialog file from a filmaker attached to his quicktime file that he thougnt was in 44.1k...was actually 48k...so when taken out of quicktime and put in another enviornment for editing that became really important and avident that the samplerate audio rules outside of quicktime applied...
Am I correct then in my understanding that when working only in the quicktime enviornment that the rules of audio sample rate do not apply?
Thanks for thoughts...
regards
vibes

All you may ever want to know about QT, audio and OS X:
http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudio/index.ht ml
QuickTime 7 is a major re-write and it now has many more options for audio. Multiple channel support (up to 24) and sample rates of double DV Stream settings.
Magic is how it works.

Multiple sample rates!!!

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