Sample Sweeping Audio Frequency

I am able to do the harmonic analyzer on a specific frequency but wanted to get input on two things:
Analyzing a sweeped signal (say an audio range of 20Hz to 20kHz) - Do most programs just sample every few periods within the file and then analyze the next?
How do programs do "real-time" analysis. I assume there is some sort of aperture time of sampling so sample every 1 second of data? In addition, I have only devised a way to analyze a wav file (converted to an array). Is it possible to do "real-time" sampling from the incoming audio?

Hi ngay528,
Essentially, you can't do a THDN analysis on a signal with a continuously changing frequency, as by definition, THDN required one fundamental frequency to get any useful information on it. In order to get proper data from a THDN the best option is to acquire your data in chunks at discrete, or stepped frequencies. If you know where these frequency changes are occurred you can then perform a THDN on those individual chunks of data.
Regards,
Hassan Atassi
NI Community Project Engineer

Similar Messages

How to get audio frequency

hello
This is sudhakar
working as a programmer on Java
now we are implementing audio recording and playing..
but here we are facing one problem..
that is .
how to get the audio frequency...that is pitch but
it variate the pitch which during the play...
if any sample code please send them..
j.sudhakar

If you want to implement an amplifier like most Players have,
that's not that easy. You will have to look for "Discrete Fourier Analysis" I think. You will have to deal with imaginary values (that is the square root of -1), if you want to write the functions up from the scratch.
Perhaps there are some prewritten function, you can use.
Good luck
Djamal

FS7 audio Frequency Response 50Hz - 20KHz?

Hello guys, I'd like to point out an issue that I couldn't find on the forum regarding the audio frequency response.I've always worked with pro camcorders that recorded 20Hz to 20KHz, which is the standard frequency us humans can hear. So I was surprised when I saw in the FS7 manual that the audio Frequency Response is 50Hz to 20KHz. Is this a typo? Or is the camera actually limited to 50Hz for the lower freqencies?I would assume that the internal mic would be limited to that range, yes, but not the internal recording of the sound? Why limit the camera to 50Hz? Any light on the subject would be appreciated. Thanks!

The tests I posted are for the mic inputs, not line inputs. I did test both, but since I often run & gun and cannot support all the extra gear to make the FS7 mic inputs on a par with Sony's EX3 mic inputs, I'm requesting that the high pass filter added with firmware 2.0 be removed in futre updates. We have the wind filter already. No additional filter is needed. While Genelec monitors are pretty good, they're not noted for deep bass response. Most studios use the ubiquitous NS10, which are only good to 80hz, which is why few people notic the deficiency. The custom built system in my screening room has a -3dB point of 7hz. I noticed that even piano sounds thin on this camera. that doesn't surprise me, since the rolloff starts at 200hz. All I'm asking for is the same quality I'm used to with my EX3's mic inputs. Only 0.5dB down @ 20hz and no thinning of the midrange. When I do large budget orchestra shoots (rare) I use 24/96 sampling, 8 channels and large diaphragm studio condensers. My recordings earned critical praise from Peter Aczel at The Audio Critic. I was a sound engineer for 4 decades before I got my first digital camera. In addition to that, I was consulted as an independent expert on infrasonics for chapter 17 of Ethan Winer's THE AUDIO EXPERT, published by Focal Press. I was responsible for the information on subwoofers and how they operate. Acoustic and speaker design share a long history with me, and my 'day job' is amplifier design, modification and repair. I'm too intimately aware of what's going on in the signal path of an audio system and when modern digital systems deviate intentionally from DAT-quality, it really bothers me. I made a big deal about this in 2007 with the Sony HVR-V1U, and Sony must have listened, because they got it right with the EX1 and EX3 cameras. But sadly, they've reverted to these games with the FS7, though not as badly as 2007.

I am having sound issues with Pottermore while using Safari 6.1. I once muted the sound and now can't unmute it even though the icon tells me it's unmated. I can hear samples of audio books in the store section. Any ideas?

I am having sound issues with the website Pottermore while using Safari 6.1. I once muted the sound and now can't unmute it even though the icon tells me it's unmuted. I can hear samples of audio books in the store section. Any ideas? I have reset Safari. Website works fine on Firefox.

I am having sound issues with the website Pottermore while using Safari 6.1. I once muted the sound and now can't unmute it even though the icon tells me it's unmuted. I can hear samples of audio books in the store section. Any ideas? I have reset Safari. Website works fine on Firefox.

SSD: for OS and apps, or for samples and audio?

I'm thinking about adding a 256 gig solid state drive; I currently have two 1TB drives. I tend to use large sample-based instruments - Play, Ivory, Vienna, Kontakt, Omnisphere, etc. - but not particularly great track counts. If I used it as the startup disk, 256 gigs would be way more than enough (because audio and video files are on other disks.) As a sample disk it would hold my most demanding and most often used libraries.
What would be the most efficient use of that SSD speed - OS and apps, or streaming samples and audio?

Thanks for the recommendation on size. I found a great answer to my question about file locations at:
http://discussions.apple.com/message.jspa?messageID=12293000#12293000
Using the advanced Options (control click the active admin account) in the accounts pane in System Preferences, one can specify a new location for the home folder. Very practical and way easier than trying to go into all the app's libraries to tell them where their files are.
Now I can just copy my home folder to the internal drive, and delete it on the SSD. After a restart, this new location will be recognized as the correct home folder.

Separate drives for samples and audio files..?

Hello all
I am moving to Macbook Pro (previous gen) with Logic Studio after years of PC (feel like I’m standing on the edge of a precipice staring anxiously down!). Will be looking to use it live in our 3-piece band to run existing MIDI Files and live triggered BFD drum samples
Have decided on getting an Echo AudioFire 8 to handle the necessary Audio outs and MIDI connections and will be replacing our rather dated Korg 05R/W General MIDI module with virtual instruments / samples libraries
I am thinking that once the Bass (probably Orange Tree CoreBass samples through Kontakt 3) and Orchestration (still wide open to suggestions on that!) are done, they’ll be exported as Audio files or perhaps ‘frozen’ in Logic to ease the load on real-time processing
Here’s the query then:
*Assuming that the internal drive will be for OS and programs only, will having the sample libraries and exported audio song files on separate physical drives from each other pay dividends – particularly when considering that the audio tracks (probably no more than two or three stereo files) will be streaming whilst BFD is triggering live samples via MIDI input..? I’m wondering if the samples and audio separate drives situation will be considerable beneficial, or given the intended application, would this approach be merely over-speccing things?*
If that is thought to be the best approach, what do you believe to be the best way of hooking these up? I plan on using the FW400 port for the Echo AudioFire 8 and have ordered a Sonnet Express eSATA Card to hook up an external drive. Should any additional drive look to use the FW800 port or should I simply tag into the spare eSATA port on the Sonnet Express Card..? My concern with the latter is that a bottleneck would occur in the Express port and defeat the whole object
I would really appreciate any thoughts on this
Many thanks
Clive

My opinions:
If you're really only talking about 2-3 audio tracks per song, having them on a separate drive is not going to make that much difference. I'd recommend just using one drive for both audio and samples for now, you can always add another hard drive later if you're getting disk errors, or if you start using more audio tracks.
If you're looking for a good orchestral library, I recommend Vienna Symphonic Library Special Edition (about $500). VSL sounds great, and this particular set is much cheaper and still gives you a good amount of instruments/articulations.
If you do go with a third drive, use both eSATA ports. Even with a little bottlenecking, you should see substantially faster performance than with FW800. The data transfer rate of eSATA is over 3 times that of FW800.
Message was edited by: jdredge

Audio Frequency Spectrum

Hi,
I'm looking for an FCE feature (3.5), or a Soundtrack feature (1.5) that allows to display the audio +frequency spectrum+ of a selection of an audio clip. My purpose is to analyze frequencies for a section of audio (e.g. a voice or a noise), identify a frequency range to fix, and modify its gain by using the standard FCE +3 Band Equalizer+ filter or the +AUGraphic EQ+ filter (or others).
I can already use these 2 filters, but without a preliminary analysis of the spectrum I have to guess the frequency ranges. Of course the best solution would be to have a dynamic real time display, during audio playback... (BTW iTunes has something like that in the top center of the main window: it's just missing precision and details about frequency values, and dBs... - not really useful).
I found something close to my needs in Audacity: the problem is that it is not integrated with FCE and that it is a static display. Or at least this is what it seems to me... I might have missed something.
Does an audio frequency spectrum display exist in FCE HD, or in the bundled Soundtrack ?
Thanks a lot
Piero
(since I didn't find any mention of this kind of display in this forum or in the Soundtrack forum - frequency or spectrum search do not return any topic - does this mean I'm looking for something that does not make sense... ? or are there other ways to do what I'd like and I described above ?)

Thanks Ian and Tom.
Yes, the Fourier Spectrum Analysis is what I need, and Sound Studio is providing it. I'm trying to figure out how to make the best use of it. In fact I hope there is a way to display it in real time, so I can see the frequency picks moving. That would be very helpful.
In any case thanks for your help.
Piero

Sample Clocked Buffered Frequency Measurement

I have a circuit board clock which I want to test. The clock should output at ~12 MHz. I'm using a USB-6343 and would like to use the Sample Clocked Buffered Frequency Measurement mentioned in the X series manual but am unclear on how to set it up. I believe I need an external clock but am not sure if I can use this card to generate this clock and what rate I should use; but more generally I'm looking for an example on how to set up this measurement in LabVIEW using DAQmx.
In the end I'd like to get an accurate average measurement of the clock frequency over ~1 second if possible and am willing to use whatever method would best work with the 6343 card.
Thanks in advance

Hi Gollum,
There's a few examples that you could take a look at that come packaged with LabVIEW. If you open the NI Example Finder (Help » Find Examples), then in the Browse tab click the following: Hardware Input and Output » DAQmx » Counter Input. In there we have a few examples on how to read in the frequency using DAQmx calls. The property that is mentioned in the Sample Clocked Buffered Frequency Measurement (CI.Freq.EnableAveraging) can be located using the following method:
In LabVIEW go to View » Class Browser, when the Class Browser window open, for Object Library choose DAQmx and for Class choose DAQmx Channel. In Properties & Methods expand the following: Properties » Counter Input » Frequency » Measurement Specifications » Enable Averaging. The Class Browser can be used to find several property nodes, and is very useful. You can play around and find various properties that can be changed in there.
Hopefully this helps.
Matt S.
Industrial Communications Product Support Engineer
National Instruments

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

Sample Rate vs Frequency

"The increments across the x-axis of the graph are controlled by the sample rate. In this example, 8000 units along the x-axis represents
one second of audio. "
Actually not. One should sample at twice the highest possible input frequency.
Renee
"MODERN PROGRAMMING is deficient in elementary ways BECAUSE of problems INTRODUCED by MODERN PROGRAMMING." Me

You see basically an FFT describes the frequencies and amplitudes of samples or events. If one undersamples one cannot hope to describe the event and that's what it's all about. If one samples at a rate less than twice the maximum frequency one cant cant
hope to describe the event because one cant be certain that data is take fast enough to describe it. I'll give you a worse case example.
Consider an electrical sine wave occurring at exactly 1 khz. The signal oscillates from +1 volts to -1 volts at the rate of one khz, Let us consider what would happen if we sampled this signal at 1 khz, We would see a constant voltage at the input although
the signal is changing sinusoidally at a 1 khz rate, If we sampled at 1.5 times the signal we'd see variations. And if we sampled at twice the maximum frequency we could be sure we're sampling at a rate which will allow description of the input.
Renee
"MODERN PROGRAMMING is deficient in elementary ways BECAUSE of problems INTRODUCED by MODERN PROGRAMMING." Me
Not quite.
When you're taking a true analog signal and "digitizing it", it will never have the resolution of the analog signal ever again.
The faster the sampling rate, the closer the approximation but the actual digital result is and will always be a square wave (not triangle wave, not sine wave, a square wave).
The faster it samples it, the closer the "stair" to the next one, thus more closely approximating the original sine wave, but it can never be completly true to the original signal.
This isn't related only to sound but to any true analog waveform being "digitized".
Please call me Frank :)
Well maybe you could do something like a Bezier curve or
Bézier Spline that would only need 4 points for each half of the wave in order to match the wave or something digitally? I don't know much about it but it seems to me you could somehow match an analog waveform digitally.
Please BEWARE that I have NO EXPERIENCE and NO EXPERTISE and probably onset of DEMENTIA which may affect my answers! Also, I've been told by an expert, that when you post an image it clutters up the thread and mysteriously, over time, the link to the image
will somehow become "unstable" or something to that effect. :) I can only surmise that is due to Global Warming of the threads.

Changing Audio Frequency/Format

Hi,
I have video with 48KHz and 32 bit audio, and an external audio source (clip mic recorded onto a digital voice recorder) with 44.1KHz 16-bit audio.
Is there a way I could change the frequency and/or bitrate within Final Cut (or elsewhere) so I can sync the video and external audio with ease?
Thanks!
Rob

if you have Final Cut Pro you have automatically have Quicktime Player Pro and if you have Final Cut Studio (from FCP5 I think) you also have Soundtrack Pro.
But iTunes can do this also for you...
Rienk

Presets - 44000 sample/second audio?

Hi! I know in Pro you can customize your own presets, but Pro is too 'hard' to work with, Elements is much easier for me. I understand you cannot create your own custom project settings, but as I just browse through them, they all have 48000 audio. I need 44000.
Is it possible to do this?
I try 48000 but it just slows down the playback.
What can I do? Any suggestions? Thanks.
Kind of a stupid question, my apologies!
Under Edit> Project Settings> General I cannot modify audio

For DVD, the output format should be either PCM/WAV, or DD AC3 @ 48KHz 16-bit.
Now, if one starts with Audio with a 44KHz Sample-Rate, PrE will convert that to 48KHz Sample-Rate, at the time of Conforming. Nothing for you to do at that point.
For a bit of background, the 44.1KHz was the CD Sample-Rate, but 48KHz was chosen for DVD, and for SACD (Super Audio CD).
Another aspect of the Conforming process, and one that the user almost never even sees, is that the Audio is converted to 32-bit, floating point, for precise editing.
If you are going to DVD, then you want 48KHz 16-bit Audio. If you need other formats, then you might squeeze a little more space out of file size, by dropping the Audio Sample-rate down to the CD standard of 44.1KHz, but not by much. Personally, I would not worry about this. Now, if you absolutely do need the 44.1KHz, most formats will allow you to choose that.
Good luck,
Hunt
PS - For DVD, you cannot choose 44.1KHz, as that is not in the standard.

HP4284A Frequency Sweep with frequency points read from a tab delimited excel or lvm file

Hello,
I am using a HP4284A, and have downloaded the file here. However, I cannot figure out what format of file it takes as input: I understand this should be a .lvm tab delineated value file, but in what order is the data? If someone could provide an explanation, or an example data file, I'd greatly appreciate that.
Thanks.
Ps. I know the linked article says that 'The frequency points are as defined for the oscillator in the HP4284A user manual.', bit I have been through the manual from end to end, and I can't find anything of the sort. The closest I've been able to find is the codgin for remote access, but that isn't very relevant as Labview and the coding (C++, I believe) used are very different.
Solved!
Go to Solution.

Sure. I wound up adding some additional features to it, though, badly. It's perfectly functional but badly coded (veteran users please don't laugh); I could probably optimize it now, having learned more, but I'm too lazy to. There's a writeup, too.
Attachments:
HP4284A Frequency and voltage sweep.vi ‏288 KB
Program writeup.docx ‏12 KB

Reception of samples at other frequency of digitization

Hi!
There are samples of a sinusoid frequency of 500 Hz received at frequency of digitization Fs = 10 кгц (I receive by means of Sine Waveform). It is necessary for me, I use these samples , to receive values of samples at frequency of digitization Fs-dFs distinguishing from 10 кгц on some value (dFs) .I need to receive values of these samples by means of function Sinc, however the sinusoid received as a result differs from a sinusoid received at the same frequency of digitization (Fs-dFs) by means of Sine Waveform. Help me, please!
Alex
Attachments:
Samples SinÑ.zip ‏24 KB

Hi Alex,
I see that you have posted a VI here to take a look at, but I am very unclear what your goal is from the information you have provided. I understand you might not be receiving your desired outputs, but it will be very difficult for me to advise any changes in logic, without a better understanding of what you need. Please provide everything you can to clarify what it is you are working with and what it is that you need from this application. In the meantime, I would suggest trying to search from within some of the resources we have at ni.com/support. Here you might be able to find one of our toolkits will provide this exact functionality. Thanks.
Jason W.
National Instruments
Applications Engineer

EHD, Samples, and Audio Recording

I will be purchasing an imac which has 2 fwire ports...My concern is that I will need the 400 port for my soundcard and use the 800 for samples/loops etc. Should I partition the "system" drive and use it for recording audio or should I put the samples and loops on it? Or, should I record audio (up to 16 tracks) to the external drive?
Your system setups and advice would be much appreciated...
have a good one
3Eo

Sadly, once you've attached a FW400 device, your entire firewire bus is going to run at FW400 - not 800 to your additional drive, as there is only one FW bus in your machine.
+Or, should I record audio (up to 16 tracks) to the external drive?+
Yes. You'll get the best performance recording audio to a separate drive.
+Should I partition the "system" drive and use it for recording audio or should I put the samples and loops on it?+
Personally, I wouldn't partition it and I'd install your loops and samples on your system drive. Partitioning, IMO, is useful for housekeeping, but it's the same physical device, so I don't think you'll see a performance increase.
If you have large sample libraries, maybe you could try a USB external. I don't use huge orchestral libraries so others may give you different advice, but I do have large drum libraries and I find USB works fine for these, but the individual samples are quite small and therefore USB works ok with these.

Sample Sweeping Audio Frequency

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