Sound Waveform Average

I'm a newbie to sound concepts and I've run into a problem. Our test set-up has six microphones set up in a reverb room. The first phase is to read from each microphone for a specified amount of time and average those readings. To accomplish this I built an array from each microphone in a while loop over the specified time interval. When the loop is done executing the average of each array is calculated. My question is how to take the now six averaged arrays and come up with one single waveform that I can use with the Octave Analysis function in the Sound and Vibration tool kit? I'm using LV 8.2.1.
Thanks.
LabVIEW 2012 - Windows 7
CLAD

Hi MeCoOp,
If you're new to SVT you might want to take a look at the LabVIEW Example
Finder (which you can find from Help»Find Examples) under Toolkits
and Modules»Sound and Vibration»Getting Started. Is there a specific part
of the application that you're having trouble with? Do you already have a code
that gives you six arrays and you are simply trying to get an averaged waveform
from them?
Best regards,
Jordan D
Applications Engineering
National Instruments

Similar Messages

Waveform averaging

Hello.
I have to acquire noisy sine signal (about 30 kHz) with NI PXI-5105 60 MS/s digitizer. I tried to make a VI using LabVIEW help but got the following result:
Apparently, it is necessary to make waveform averaging but I don't know how to do such a thing.
Here shows how signal is indicated:
I found here the following example from RavensFan, but could not use it:
There is a type mismatch:
Is there somewhere a detailed description of the LabVIEW's waveform / complex waveform / … data types with examples? I so stupid and standard LabVIEW help did not help me.
Thanks in advance for your help and sorry for my poor english.

Try:
Extract Single Tone Information VI
Owning Palette: Waveform Measurements VIs
Requires: Full Development System
Takes a signal in, finds the single tone with the highest amplitude or searches a specified frequency range, and returns the single tone frequency, amplitude, and phase. The input signal can be real or complex and single-channel or multichannel. Wire data to the time signal in input to determine the polymorphic instance to use or manually select the instance.
and
Averaged DC-RMS VI
Owning Palette: Waveform Measurements VIs
Requires: Full Development System
Calculates the DC and RMS values of an input waveform or array of waveforms. This VI is similar to the Basic Averaged DC-RMS VI, but this VI gives more precise control over the individual DC and RMS calculations.
Do you really want the RMS or the amplitude of the 30kHz signal??
Have a look at the (power) spectrum of your signal .. will help you identify (and eliminate) the noise.
Greetings from Germany
Henrik
LV since v3.1
“ground” is a convenient fantasy
'˙˙˙˙uıɐƃɐ lɐıp puɐ °06 ǝuoɥd ɹnoʎ uɹnʇ ǝsɐǝld 'ʎɹɐuıƃɐɯı sı pǝlɐıp ǝʌɐɥ noʎ ɹǝqɯnu ǝɥʇ'

Universal upgrade, Viewer not displaying sound waveforms

That's pretty much it.
New macpro, universal upgrade.
All is okay (apart from the odd quirk) but I cannot see waveforms in the view, either opening them from the browser or the timeline.
Any ideas?
rs

Rosetta-
I have a similar problem as you. However, mine is going from an Intel Mac to a G5, and the AIFF waveform is not showing up on the G5. This bug was not happening until I upgraded to FCP 5.1.2 today. Here is the question I posted in another thread:
I've encountered a strange bug after upgrading to 5.1.2 today. I have 3 Macs with FCP (all upgraded to 5.1.2 today). Was working on a project at work on one of them today and everything worked fine. I imported an AIFF file and decreased the levels on it as part of the project.
When I brought my project home (I use hot swap hard drives), I went to adjust the levels of that same AIFF file within my project. However, the viewer pane went half black, it did not show me any bit of the audio file and it's peaks. However, I could still adjust the levels. It just seemed to be that there was some sort of video display bug when I tried to adjust the AIFF file in the viewer.
I looked at some other audio files (recorded through the voiceover control) and they appeared fine in the viewer. Only AIFF files gave me this bug. To try and remedy the problem I tried other AIFF files and also tried restarting the project, etc. Nothing cleared up this problem though. Anyone else experience this?
If you have any luck with this, please keep us in the know.
Thanks,
Ted

Want to buy an ipod shuffle..Does it sound good with the earpods??Asking because my earpods sounds pretty good and loud with the iPad but sounds very average with my Macbook Pro 13 non-retina

............

"Good" is a personal opinion, when it comes to audio gear. If there is an Apple Store nearby, you should take your EarPods there and ask if you can connect it to one of the display shuffles, and try it out.
In my personal opinion, the EarPods sound very good in general.

Synchronising recording of eeg waveforms with stimulus

Hello everybody ,
im working on a VI to average several EEG waveforms ...which is attached ..
actually each waveform entering this ' waveform averaging' VI is actually an EEG reponse to each sound click we present to the subject.
so i need each click to trigger or start the recording .. and continue recording for 17 milliseconds.
Thus if i give 1000 clicks, the averaging VI must produce the average of the 1000 waveforms.
the clicks are given as a square waveform , frequency = 10.1 Hz , TIMING mode: simulate acquisition timing , samples per second = 44000 , no : of samples = 204000 (or a figure as high)
i used the simulate signal and play waveform blocks.
for the above problem, i tried to use the trigger and gate block... i thought of connecting the simulate waveform block to the waveform averaging vi using the trigger block...but dont seem to reach anything useful ...
P.S. : this block diagram has sinusoidal signals as the example .. in reality we record EEG signals .
I wud really be grateful if u cud help me out ...
best regards
Mer
Attachments:
AverageWaveforms2.png ‏6 KB

Hi, I'm not LV expert but I think I run into some thing like yours before. What I did was checking any missing pulse or abnormal in the EEG waveform and set trigger pulse once I found it. Take a look at your signal amplitude. If your signal too low (ex: <1.25V), your counter gate won't trigger. Once you record your EEG, amplified it and use trigger VI (take a look at the LV example) to set your trigger event pulse.
BC

No Sound on R3D files in AE

I have some R3D files here from my Red Scarlet and Epic. I used the A Box from wooden camera and a microfone with xlr to record sound direct on footage. In Redcine X and Premiere i got the waveform and the sound, but in After Effects i got only a dead spine. No visible waveform. I can activate sound button on and of, also can change the DB, but nothing happend. I also got no sound with Ram preview or rendering. If i put the footage in a Premiere project and import this Pr Projekt to after effects, than i got the waveform. But that is not the way i want to use my footage. Now i lost al the RAW abilitys from my R3D files and some other specials
Does somebody know, why i got no sound/waveform?
I have the latest CC version of AE.

I got an other project now. First time i imported the files, i got al my waveforms. Than i start working. Deleting, cut, duplicate, some files in the composition. But i did not deleted the files in the project. I leaved them al imported. Since i have done a bit of cuting, i lost al my waveforms. If i add al files to another comp i also have no waveforms. Reload footage has no effect. Also starting a new project and import the footage has no more effects on the waveform. I do not get the waveform anywhere.
Pls i need some help, why i lose my waveforms

Realistic drum sound? Advice given and needed

I am trying to figure out how to get a decent, realistic (ish!) drum sound on the iPad, to use when recording garageband tracks. Let's face it, the smart drums just don't cut it. I have gone through seemingly tons of set-ups and configurations with different apps and so forth, and have sort of settled on a sound which is ok (for now) and is by far the best sound I've achieved so far. Bear in mind that, as a garageband user, I am not investing tons of money into tech - a few pounds per app is all I've spent really. First, I'll describe what I've done in the past, and what I currently do.
I started with the smart drums, of course. Tiresome very quickly, and they were naturally holding back the quality of my recordings by making them sound, basically, rubbish. I find that with drums, they either need to sound decently realistic, or completely electronic (like a drum machine). Anything in-between is nonsense and sounds bad.
I next downloaded the DM1 app and made drum tracks using some of the preset acoustic drum kits. Much much better sound, ability to change individual drum sounds and panning. Still not great though, and very laborious as the track had to be (before ios7 and inter-audio apps were usable with garageband) rendered, emailed to myself, downloaded, converted using file conversion software, loaded into another DAW to allow it to be copied, then pasted into garageband. There was also some latency apparent when exporting the file. Bah!
Eventually I found Easybeats 2. More intuitive, some ok-sounding samples available to use for the different drums. Only 16 bars of drums can be programmed at a time, but that doesn't matter so much as you can easily export the 16 bars into your pasteboard straight from the app, then copy it into garageband (no latency either!). Then just wipe the 16 bars and do the next section. Panning, individual drum control, VELOCITY CONTROL for each hit (finally!).
Still something was missing though. I had yet to find a kick and snare sound which was close to acceptable. DM1 came closest for those two so far, but much worse for cymbals, hi-hats, toms, and had no velocity for each hit (well I think you could select normal or louder for each hit, but nothing more than that - not good enough). Also the snares were a bit weak-sounding. For some time, my solution was to record a drum track on kick and snare only using DM1, load this into garageband, then record a duplicate track in Easybeats 2, again with kick and snare but this time with everything else as well, then load that on another track, and attempt to sync the tracks as closely as possible (sometimes a laborious job considering the latency from DM1), then mix this down to one track. This gave a powerful sound and I was pleased with it for a while, though looking back now, it was awful. Just awful.
When I became dissatisfied with this sound, I came to a crossroads. Was I going to stop this quest for realistic ipad drums, and just buy EZdrummer for my laptop? I would need to upgrade my windows as well. And buy a DAW. An investment of hundreds of pounds! No, I would continue my noble quest! But for this, I would need to enlist the help of my laptop.
What I really needed was a decent kick and snare sound to use in Easybeats 2 (i tried Easybeats 3 but went back to 2. 2 is just better in my opinion). For me, that's 80% of a good drum sound. Average-quality cymbals and hi hats can hide to some extent if the snare and kick are convincing enough. Of course, I'm talking about rough-quality demo tracks - in professional recordings all elements are of course important. I would have to find samples, and save these as a drum kit to be used with Easybeats. But this can't be done using only an ipad. I googled 'acoustic drum samples' on my laptop and on the first page is 'Tom Hicks Samples'. Zip files with tons of samples for every drum you could need, and all free. I don't think you can find better for free online. Still I was not 100% happy with any one sound. I therefore loaded a few sounds into Audacity (free mixing software) and overlayed them, and played with the bass and gain, until I was happy. I then exported these as wavs, created a custom wav kit by sending a compressed file with all of my desired sounds from my laptop directly into Easybeats 2 (you need to plug the ipad into your laptop to enable this), and finally... Finally! My drum kit is completely (for now). The combination of decent sounds and the ability to control the velocity of each hit in Easybeats 2 is, thus far, simply the best method I've discovered for drums on the ipad.
Not being very technically-minded (I prefer just playing and writing music, but know little about tech or computers) the above methods were things I discovered and experimented with gradually, and something as seemingly simple as, say, creating a custom snare sound by overlaying multiple samples into Audacity, then using that sound for every snare hit, is not something which even occurred to me for a long time. There was certainly nothing online that I found to explain or guide me through any of the steps. Most advice amounted simply to 'use DM1', or to not use the ipad at all. Hopefully my advice may prove useful to anyone undertaking a similar quest.
I posted this not just to explain my method, but to ask whether anyone has any input or further advice on creating a good drum sound on the ipad. I am always looking for new ideas, and I am not convinced that I have reached the pinnacle of what the ipad can achieve yet.

I'm definitely going to agree with BeeJay to a point... Logic's drums are not stellar (they're okay, but that's as far as I go), and forget Ultrabeat if you want a little bit a "realism". However, if all you have to work with right now are Logic's kits (and perhaps the Rhythm Section JamPack), go into the EXS and separate the important groups (kick, snare, toms and hi-hats) into their own aux channels and save the the instrument as a multichannel instrument. Work with the separate aux channels by adding in a bit of EQ and/or compression (or do the New York style mix). As well, send the unprocessed signals to a bus containing a drum room reverb and mix them together.
Another thing you may want to consider is the fact in order to sound like a human, don't play it like a robot. Quantize certain parts, but not every part (and it doesn't have to be at 100% - use lower strengths to bring out a bit of humanness). If you don't have one, get a pad input, such as M-Audio's Trigger Finger. Note the a lot of the kits in the EXS and JamPacks have a few articulations for drums that you need to make use of, such as flams, drags and rimshots to do various beats. Listen to various rhythms and fills and watch how drummers do it (plenty of videos on http://www.drummerworld.com). And, when you make a beat, play the beat.
Back to samples: if you find that Logic's kits still aren't cutting it for you, definitely go for something like BFD, DKFH or whatever. I fell in love with BFD and highly recommend it. As to whether it sounds like a drummer for you will depend on how you play it. For me, it definitely sounds like a drummer. :-D
jord

Waveform fuzzy recognition?

Sorry,I really don't know if my question belongs in this category.Please see the attached file.There are 105 data points and they are plotted as the waveform.
It is not such regular waveform as sine/square/triangle
.From eyes we can see there are three peaks and three valleys respectively in the waveform.I wonder
if the computer can do this.There is a "peak and valley detection" example in labview and it does work to some extent.But if there are some pseudo-peaks(valleys) it does not work as well as what I expected.
Can Matlab or any other software do this job?Is it related to the knowledge of signal processing(wavelet or ...??) Could anyone give me some helpful hints?
Thanks in advance.
Attachments:
waveform.jpg ‏53 KB

Greetings!
I understand your question fine There are a couple of things you want to do. First, you might just try a different INTERPOLATION. The graph pallette gives you a few choices in this regard, the default is linear. If that doesn't work sufficiently, you want to try waveform averaging. Depending on how many points you average, you can eliminate "local" peaks while retaining longer peaks. There is a Waveform Average vi. in the NI code archives..(I think it's the one I submitted myself a while back!) Keep in mind that when you average a waveform, your peaks will be offset by half of the averaged samples....no way of gettin around this without "cheating". Hope this gives you some ideas. And yes, this can be done in Matlab quite nicely, and t
here is probably already a waveform average in the Matlab Tools.
Goood Luck
eric
Eric P. Nichols
P.O. Box 56235
North Pole, AK 99705

Visual "amplification" of the audio track

I'm running Premiere Elements 10 and find the visual representation of sound in the audio track very useful. However, the only sound that disturb the horizontal line are relatively loud. Other quieter sounds - particularly speech - often cannot be "seen". Is there anyway of widening the audio track on the screen to get a more detailed view of the sound waveform - I would find it very useful when editing?

There are 2 settings for all track heights. Click on the icon at left of the track to change.
Try 'normalising' the sound. Sound waveforms should fill the track height only where there is a loud peak but normal sound should be about half height.
If the sound "waveform" is very low, it means the original material is recorded too low. Maybe you have a fault in your camera?
Maybe the camera has been set for loud sounds or it's auto volume system is not working properly or not switched on,
Cameras sometimes have a "boost" setting for soft sounds. Try this if so
Sometimes a faulty electret mic has peaks going in one direction, this halves the average level (and causes some distortion)
The average level of speech at the normal level should be about 6 db below the loudest peak (spike in the waveform).
Sometimes a lound intermittent noise can force the rest of the sound in a clip down due to the auto level system in any camera.
You can restore the level of any clip by "normalising" it. (Selectable from the menus after selecting the sound clip to be normalised)
This sets the LOUDEST peak in any clip to max level and if this peak is already lower then the maximuim, then brings up the whole clip as well.
If an offending spike, such as a bumped camera is in the way, you can zoom in on the spike, sets cut points either side of it and reduce just the level of the spike by say 6db, then normalise the whole clips either side of it to bring it all back up to normal.

Need help in buying a Laptop

Hi. I am looking to buy a laptop to replace my desktop.I will be using this laptop in my home only( room to room).For surfing the web,e mail,watching you tube videos,Facebook.And one last important thing,playing on facebook poker games.I think it"s called Zanga on facebook.I remember my old laptop before i bought my desktop,it use to freeze up sometimes while i was playing on Facebook-Zanga poker games.I will not be using my laptop for any other games but poker games.My question is do i need a special graphics card or can i buy any computer for these things .I am 60 yrs. old and don"t know a lot about computers even though i have been using them for years.I can not go more then $600.00.Please help me decide.Thank you for your time in reading this.

Sounds pretty average user requirements. As long as you don't plan on running any graphic intensive software or doing anything like mixing audio professionally you can probably grab whatever you want. Sorry I don't have any other advice for ya. Go see what the blue shirts say, that's what they're there for and they can get a better sense of your needs in the store and actually show you different products. It will also give you an idea of screen size, you'll find a keyboard that you can deal with, sound quality (for what it's worth at least).
$600 is a nice chunk of change to put into a computer now-a-days so I doubt you'll have trouble find anything.
I understand some people in your age range like to bring in family members, but I would urge against it since what happens your relative although they come with good intensions tend to look for stuff that they want, not you.
Example- you may want to end up looking into a tablet, like an iPad or Microsoft Surface tablet, with an attachable keyboard (look up a commercial on youtube if you need to see an example), but your son-in-law might tell you to get a 17 inch laptop that ends up needing plugged in all the time.. Have you ever sat at a Starbucks comfortably with a huge and heavy laptop and close to an outlet.... I haven't.
Former BBY Home Theater Associate 2010 - 2012. Now I work as a graphic designer in the print and prepress industry.

Neo 2 865PE Setup Options-Opinions Wanted

Hi, I finally got my new PC set up and I have some setup questions I'd like to get some opinions on:
1) My Hard drive is an ATA133 from Maxtor. Is it better for me to use the IDE1 slot or the Promise IDE3 slot? If I use the IDE3 slot, I will be able to run all of my IDE using peripherals as masters and I wont have to slave anything.
2) Sound- Is the onboard sound better then a Creative Labs Soundblaster Live card?
3) Bios Settings....what are the best performance settings I should use considering the following hardware:
- MSI Neo 2 FISR 865PE Motherboar
- 2.6GHZ 800FSB P4 CPU
- two 256MB Giel 533MHZ PC4000 RAM (512MB total)
- Gainward Ti4200 G4 128 MB "golden sample" video card.
I read in Toms Hardware in this article that I should be able to Overclock the BUS to 900 and perhaps 1000 with a setup like the one I have...the only major difference is that I got 512MB of RAM where in the article he uses 1000 MB of RAM.
http://www.tomshardware.com/motherboard/20030812/index.html
Any comments or suggestions are appreciated.

There'll be no difference of ATA100 and ATA133, the speed is not easy or almost impossible to be detected on daily usage.
I'm pretty happy with my NEO2's onboard AC97' chip, it delivers good quality sound for average users like me, regardless in terms of gaming, musics or movies. Unless you want very high quality sound likewise DTS, THX or doing music ripping, then you may need a better sound card like Audigy 2.
Your setup is ok, but be very careful with your RAMs as the NEOs are not yet very stable at support RAMs of more than 400MHz in speed, during the setups, if you're encountering instabilities, I'll recommend you to lower your RAMs speed to 400MHz and lower all latencies and increase the RAMs Voltage to 2.7v, set AGP/PCI Freq to 67.01/33.50Mhz.
All the Best... :D!!!

Video clip images have disappeared in the timeline

Hi - I was working away on an event in FCPX when I noticed that the timeline no longer showed any of the clip images - just the sound waveforms. The video is visible on the canvas, and can be edited, but there is just a segmented blue nothing where the clip images used to be. This is a new one - anyone experienced this before, or have an idea what's causing it?
Thanks!

Thank you very much, Tom, for your very clear reply! It worked using the Option-Command reset of preferences. Interestingly, when I clicked on Close to shut down FCPX, this is what popped up. I have no idea what "background tasks" were being undertaken. Although the app disappeared from the screen, it remained live in the dock and had to be force quit. Do you think it's possible that some not-deliberately-invoked process removed the clip images, and if so, what it might be? I can trip on a bare floor, but don't usually send an app off accidentally!

Plsql datatype -- URGENT HELP..

Is there a plsql datatype i can declare which can hold > 32k bytes value.
please note that data value is not coming from database.
rather i will be getting it from an OUT variable of an external program call from plsql stored procedure.
THANKS.

LOB Types
The LOB (large object) datatypes BFILE, BLOB, CLOB, and NCLOB let you store blocks of unstructured data (such as text, graphic images, video clips, and sound waveforms) up to four gigabytes in size. And, they allow efficient, random, piece-wise access to the data.
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LOB types store lob locators, which point to large objects stored in an external file, in-line (inside the row) or out-of-line (outside the row). Database columns of type BLOB, CLOB, NCLOB, or BFILE store the locators. BLOB, CLOB, and NCLOB data is stored in the database, in or outside the row. BFILE data is stored in operating system files outside the database.
PL/SQL operates on LOBs through the locators. For example, when you select a BLOB column value, only a locator is returned. If you got it during a transaction, the LOB locator includes a transaction ID, so you cannot use it to update that LOB in another transaction. Likewise, you cannot save a LOB locator during one session, then use it in another session.
Starting in Oracle9i, you can also convert CLOBs to CHAR and VARCHAR2 types and vice versa, or BLOBs to RAW and vice versa, which lets you use LOB types in most SQL and PL/SQL statements and functions. To read, write, and do piecewise operations on LOBs, you can use the supplied package DBMS_LOB. For more information, see Oracle9i Application Developer's Guide - Large Objects (LOBs).
BFILE
You use the BFILE datatype to store large binary objects in operating system files outside the database. Every BFILE variable stores a file locator, which points to a large binary file on the server. The locator includes a directory alias, which specifies a full path name (logical path names are not supported).
BFILEs are read-only, so you cannot modify them. The size of a BFILE is system dependent but cannot exceed four gigabytes (2**32 - 1 bytes). Your DBA makes sure that a given BFILE exists and that Oracle has read permissions on it. The underlying operating system maintains file integrity.
BFILEs do not participate in transactions, are not recoverable, and cannot be replicated. The maximum number of open BFILEs is set by the Oracle initialization parameter SESSION_MAX_OPEN_FILES, which is system dependent.
BLOB
You use the BLOB datatype to store large binary objects in the database, in-line or out-of-line. Every BLOB variable stores a locator, which points to a large binary object. The size of a BLOB cannot exceed four gigabytes.
BLOBs participate fully in transactions, are recoverable, and can be replicated. Changes made by package DBMS_LOB can be committed or rolled back. BLOB locators can span transactions (for reads only), but they cannot span sessions.
CLOB
You use the CLOB datatype to store large blocks of character data in the database, in-line or out-of-line. Both fixed-width and variable-width character sets are supported. Every CLOB variable stores a locator, which points to a large block of character data. The size of a CLOB cannot exceed four gigabytes.
CLOBs participate fully in transactions, are recoverable, and can be replicated. Changes made by package DBMS_LOB can be committed or rolled back. CLOB locators can span transactions (for reads only), but they cannot span sessions.
NCLOB
You use the NCLOB datatype to store large blocks of NCHAR data in the database, in-line or out-of-line. Both fixed-width and variable-width character sets are supported. Every NCLOB variable stores a locator, which points to a large block of NCHAR data. The size of an NCLOB cannot exceed four gigabytes.
NCLOBs participate fully in transactions, are recoverable, and can be replicated. Changes made by package DBMS_LOB can be committed or rolled back. NCLOB locators can span transactions (for reads only), but they cannot span sessions.

Dragging music from iTunes not working

When I drag a song from iTunes to the Timeline there is no sound (waveform). The file plays ok when I click it in the Music tab >Itunes > Music folder
Colin

Does the song have DRM protection?

Why is my iPod transferring dreadfully slow...

No, I haven't ever used FireWire, aside from working with DV cameras, but as far as I can see, my iPod is only transferring at ~12 mB/s. However, the USB2.0 specification allows transfers up to 60 mB/s. So, how come my iPod isn't syncing at full USB 2.0 speeds? My computer is decently equipped, so...
And, out of curiosity, will there be an eSATA version anytime soon? =)

Why did you bring up Firewire? We are talking about USB2 right?
Anyway, the theoretical transfer rate of 480 Mb/s (60 MB/s) is just that, a nice theory that will almost never be reached and certainly not at any sustained time frame. You would be very lucky to even hit 300 megabits per second and then probably one for short peaks/burst. Sustained transfer speeds are likely to be much lower on average. My guess is you would be lucky to see average speeds of 100-150 Mbits/s (12.5 MB/s - 18.75MB/s) if you had a good system with a good match up of components connected via USB2.
So in other words, my opinion is your 12-ish megabytes per second sounds about average.
Cheers,
Patrick

Sound Waveform Average

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