Audio sample at variable frequency

I think this is the right forum for that question.<BR>
Do anyone know if it is possible to play an audio sample an modify the frequency while playing according to a variable value?
Any example?

write yourself a Mixer that can change the frequency of the data flowing through it.</p>
Mixer
<p>
Or find one on the internet that does it.
<p>
matfud

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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!

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.

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 compress an audio sample?

Hi, I have raw audio sample in a byte array. I want to compress this sample. more specifically, I want to develop a method which will take this byte array as an input and returns the compressed sample in the byte array again. Please suggest me if anyone knows about this.
The code which I am using for capturing the data through the microphone is as follows. After executing the following, a byteArrayOutputStream is created, from which I am getting the captured audio byte array just by calling byteArrayOutputStream.toByteArray() method.
public void captureAudio(){
    try{
      //Get everything set up for
      // capture
      audioFormat = getAudioFormat();
      DataLine.Info dataLineInfo =
                new DataLine.Info(
                  TargetDataLine.class,
                   audioFormat);
      targetDataLine = (TargetDataLine)
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                         dataLineInfo);
      targetDataLine.open(audioFormat);
      targetDataLine.start();
      //Create a thread to capture the
      // microphone data and start it
      // running. It will run until
      // the Stop button is clicked.
      Thread captureThread =
                new Thread(
                  new CaptureThread());
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      System.exit(0);
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class CaptureThread extends Thread{
//An arbitrary-size temporary holding
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byte tempBuffer[] = new byte[10000];
public void run(){
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}//end inner class CaptureThreadI am new to sound api and I hope you got my question. Please ask me if anything required from my side.
Regards

Thanks Andrew and Captfoss, I agree both of you and I am really a newbie. Andrew, you are talking about the decreasing the quality, like what we do in video conferencing application where quality of an image not really matters [we set the jpeg quality to 0.5 etc]. I'll definitely do that. Please tell me the values for lower quality sound. The audio format I am using currently is as follows:
private AudioFormat getAudioFormat(){
    float sampleRate = 8000.0F;
    //8000,11025,16000,22050,44100
    int sampleSizeInBits = 16;
    //8,16
    int channels = 1;
    //1,2
    boolean signed = true;
    //true,false
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    //true,false
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                      sampleRate,
                      sampleSizeInBits,
                      channels,
                      signed,
                      bigEndian);
}//end getAudioFormatcaptfoss, I also thought of ULAW compression. But I am not getting the way to apply this compression on the piece of sound which is stored in a byte[]. I am capturing the sound in piece by piece through the microphone.each piece of sound is stored in a byte[]. Now I want to compress this byte[] using ULAW. How can I do that? I don't have any file on the hard disk as I am capturing through the mic. Also, I don't want the compressed file to be stored on the hard disk. Instead, I'll want the compressed output in a byte[].

IMac hard drive making clicking sound (audio sample)

A couple of nights ago I realized my five year old iMac was making a sound I have never heard it make before. It is a scratching noise that I assume is coming from my internal hard drive (hard drive is stock, has never been replaced). It isn't a super loud noise, but it sounds like the noise some hard drives make when they are "searching" or "thinking". I'm wondering if my hard drive is beginning to fail, or if the problem could be something as simple as dust or dirt caught in the drive. An audio sample is in the link below, please help me diagnose my problem. Thanks!
Audio sample: http://cl.ly/073d3U1e1t280M2X0m3F

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AUDIO SAMPLES FROM IOMEGA 100 ZIP DRIVE TO MAC BOOK PRO / LOGIC?

Posted: Fri, Feb 15 2013, 1:27pm Post subject: AUDIO SAMPLES FROM ZIP 100 TO MACBOOK PRO / LOGIC?
HI THERE
I HAVE LOADS OF OLD AUDIO SAMPLES ON ZIP DISKS ( ZIP 100 using akai cd3000 sampler ) THAT I WOULD LIKE TO TRANSFER TO MY MAC BOOK PRO FOR LOGIC EX24.
IS IT A SIMPLE CASE OF A 'SCSI TO USB LEAD' , WILL MY MAC BOOK PICK IT UP FROM THERE OR DO I NEED SOFTWARE OR DRIVERS FOR MY MAC TO BE ABLE TO SEE THEM?
IS IT EVEN POSSIBLE AT ALL?
THANKS

Hi guys
Sorry for the delay in coming back. I bought a zip drive pretty cheaply then and it has only just turned up. However when I put my old ZIP disks in, it seems like the ZIP drive reads them at least but the MACBOOK comes up with the following message...
'The disk you inserted was not readable by this computer'
Does this pretty much mean a no go then? Or can I put the samples from my AKAI sampler back through the old ZIP drive, reformat some new disks in a certain way on that drive so that my MAC can read them?
All its for really is mainly to take old tracks that used these samples and getting them set up in my mac. Also there was some good bits and pieces there anyway and would like to keep using those samples for newer projects. I could of coourse resample them from the AKAI straight to the MAC but would be not as practical and would take alot longer. The 13'' macbook pros input is not digital, only its output.
thanks

Sampling local variable and synchronize with DAQmx

Hello,
I made a small change in the set-up I used with labview and now when I wanted to change the code I'm having a rather complicated problem.
In my old set-up I was measuring three variables: x and y with a QPD and the power of a laser with a power detector. I was using the DAQmx and I was getting a matrix with three columns with n (sample rate) values. Now, for various reasons I had to take out the second detector. So now I want to build the same matrix as constructed before, but instead of putting the measured values of the laser power I want to put the theoretical values (they are in a local variable) as I cannot measure them. The problem is that this local variable, in general, changes during the DAQmx acquisition time and I would need to sample it at the same rate as I acquire the data from DAQ and then combine all them. How I could sample this variable and attach it to my DAQ results? DAQmx doesn't accept local variables.
Thanks

A local variable is not something standalone. It is always associated with a control or indicator. Hows is it updated?
From your description, it is not clear what you are doing. Can you show us some code instead?
(Also be more clear when using acronyms. QPD cound mean many things)
LabVIEW Champion . Do more with less code and in less time .

Please Help. How do I assign a short audio sample to a specific note?

Hi. Happy new year to all!!
If any one can please please offer advise it would help to get my new year off to a much better start.
I am wanting to assign/map individual audio samples to individual 'note' or midi events, same kind of set up as how say any of the drum sets work in ultrabeat.
Is it something to do with ch assignment?
The result I'm looking for is if I hit C0 on my keyboard, I get one sample. If I hit G4 on my keyboard, I get a different sample and so on.
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I have searched extensively in the manuals and the discussions and suspect the 'way' I'm thinking about this might be wrong. I'm sure it must be possible.
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Hi Thanks for that.
Since my posting, I discovered another option, by using the Sampler exs24 and creating a new key map, this seems to be exactly what I was after. Like all things, its straight forward when you know how.
I looked into the ultra beat sampling a bit further as I understood it, before seeing your reply, to discover that the sampler window was about chopping/or linking a sample to a preexisting rhythmic sequence. This was not quite what I was after but I'm interested to look further in to your suggestion which seems a bit of a different thing again.
Where theres a will theres a way and I googled my question leading me to the macprovideo discussion group with the following link:
http://www.macprovideo.com/forum/logic/logic-pro-express&id=5356
so this was enough to send me in the right direction.
Not being inundated with answers form the forums, your responce is very appreciated.
Thankyou.

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..
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Thanks:)

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

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i have logic pro 8 and i was wondering how to send chopped up samples of an audio file to my midi controller which is an axiom 49. for example, i wanna chop up some vocals and instrument loops from a song called dazed and confused to create a hip hop song and i ve seen it done before on youtube just wondering how to do it.

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You'll need a basic familiarity with editing in Logic, as well as some fundamental knowledge of the EXS sampler or Ultrabeat.

Can you apply varispeed to specific audio samples in a session?

Hi all,
Is it possible to apply varispeed to specific audio samples in a session?
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If you want to apply an effect to everything in your session, then open the mixer and put the effect into the master channel. It's that simple...

How can I ensure that the audio sample rare of my capture preset matches?

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Hi
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Hi all!
I wanted to output a signal via the soundcard with one variable frequency.
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ANDY

Hi Andy
I'm afraid I've only got access to 7.0 & 7.1.
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Hmmm...
I've had a scan around and you could try checking this link
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HT201808 "...nor may they be repackaged in whole or in part as audio samples, sound effects or music beds." So does this mean I can't make a song with just remixed loops and publish it on SoundCloud for example?

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momomikes,
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https://support.apple.com/en-us/HT201808
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Audio sample at variable frequency

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