AA3 Sound input from computer mic port-help needed

My new HP desktop has Vista 64 home premium (with creative soundblaster soundcard upgrade) and
I've installed audition3 and have it working fine when using audio files
but now I want to record audio using the mic port on the desktop. I have a headset mic (from dragon speaking software)
and the windows sound recorder hears the mic just fine. I am going nuts trying to get AA3 to hear it!
Any turotials I haven't found yet?

I spent 3 hrs at it, durin and steve thanks! now
there is audio from the mic input in edit view. I hope to find it in multitrack view but need a break!
my problem with vista is that i'm so anti-change that i'd rether stay with it. I dont know how to change to Win 7 and as Steve says there3 is concern that we waqit for next version so the bugs are fewer. like Steve, I liked XP; I would take it back, but I'm not a tecchie and have no time to learn these thiongs.
thanks again
JohnO

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"Using the harmonic analyser to determine the musical note of sound input from a mic and soundcard"

"Hi there, I am currently trying to extract what musical note is being sung via a mic plugged into my soundcard and using the harmonic analyser VI, I am experiencing problems however, as the fundamental frequency output does not always match the note being sung. Does anyone have any ideas why this may be? I initally thought it may be due to aliasing occurring but this is compensated for in harmonic analyser VI! If anyone has any suggestions I would be extremley grateful. Thanks
Em"

[posted and mailed]
em wrote in news:5065000000050000009F680000-1012609683000
@exchange.ni.com:
> I'm pretty sure it would work fine with pure tones (although I do
> intend to test this), my problem really is how to fix it so that it
> can work with voices! One idea I thought about was to apply the
> A-weighting to the waveform to more accurately model the way a human
> would hear the sound, but I'm not sure how much difference this will
> make, any other suggestions would be very useful.
> Thanks
> Em
Rob Cole is right, your question is one that has a lot of potential
answers.
Here are a couple of notes / thoughts:
A-weighting probabably won't help much for your particular app, as it
sounds like your are more interested in tone detection, rather than
modeling how a human would hear or compliance with a standard.
Probably the most likely reason you are seeing the fundamental frequency
vary is that you are measuring a real-world signal that is a human voice.
Even though your singer may be singing a constant note, the frequency
content will vary somewhat--your singer is an imperfect boilogical
system. (Nothing against your singer in particular--no signer is
perfect!) Also, as Rob Cole pointed out, your singer's note isn't a pure
tone at all, but a collection of harmonics that you perceive as a note.
Regarding aliasing, it's important not to confuse any algorithmic
compensation (applied after a signal has been sampled) with true alias
protection. If you are using a sound card, it likely doesn't have
hardware-based alias protection. The only way to ensure that aliases
don't intrude on the frequency span of interest is to apply a hardware
filter prior to sampling your signal. Such filters can be added to the
front end of a DAQ board or come built-in with DSA hardware such as the
NI 4472. The software alias-related functionality of the Harmonic
Analyzer VI's simply excludes signal content that exceeds the Nyquist
rate in the distortion calculation.
I'm not sure specifically which VI you are referring to, but the SINAD
and Harmonic Distortion Analyzer VIs from the waveform analysis palette
use a tone detection algorithm to find the fundamental which is the same
as the one found in the Extract Single Tone Information VI. All of these
VIs include an advanced search input where you can specify a guess for
the frequency of interest and a range (as a percent of the sampling
frequency). You might try to narrow the search from the "try everything"
mode (-1 in the approx freq.) to something with less span around the
expected freq range.
Take a look at the power spectrum of your signal and verify that you see
a distinct, time-invarient peak where you expect your fundamental. My
guess is that you don't. I've whistled into a mic enough to know.
A final suggestion is to increase the duration (# of samples) acquired in
your acquisition. When you do so, you increase the duration of the signal
window that you analyze. If your signal shows slight frequency variation
over a given period of time, they can tend to average out with increased
duration of the signal that you analyze.
Hope this helps!
--Sam

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does anyone know of an easy way (or just a way) to use the sound input from another networked computer to mine into garageband?
thanks

Hi Christoph,
it doesn't seem to work like that (no unnecessary D-A-D conversion), as shown on page 6 of the documentation.
More information here, but I must admit that I'm no audio engineer so I don't fully understand how it works!
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hello all i have hp pavilion dv9500 and i have no sound no webcam no mic please help
on the sound i can see the green part on the volume control move so i know drivers are fine but i cant hear it at all i bought new speakers and fitted them still no sound im thinking hardwear prob there is a chip that feeds them all and also the power button could it be this please help thanks all

Hi,
barney74 wrote:
hello all i have hp pavilion dv9500 and i have no sound no webcam no mic please help
on the sound i can see the green part on the volume control move so i know drivers are fine but i cant hear it at all i bought new speakers and fitted them still no sound im thinking hardwear prob there is a chip that feeds them all and also the power button could it be this please help thanks all
Which operating system is installed?
What do you mean by "fitted speakers"? Do you mean plugged the connector in?
What is the exact product name/number of your notebook?
Is there no webcam or mic internally installed or do you mean that they are not working?
Best regards,
erico
****Please click on Accept As Solution if a suggestion solves your problem. It helps others facing the same problem to find a solution easily****
2015 Microsoft MVP - Windows Experience Consumer

Hi, I use Notion 4 like rewire slave in Logic, but I don ´t know, how to activate audio input from it, can anybody help me? Thanks, Mario

Hi, I use Notion 4 like rewire slave in Logic, but I don ´t know, how to activate audio input from it, can anybody help me? Thanks, Mario

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This probably isn't going to sound very helpful, for which I apologise in advance, but I'm afraid it's the truth...
The only way you stand any way of doing any of this is by using ASIO4ALL, which can aggregate different drivers from multiple devices. But I don't rate your chances too highly, I'm afraid - USB mics have long been the problem child of computer-based audio. On top of that, you are trying to get sounds in and out of a separate device at the same time - not only that, but it's a Creative Labs gamer's device and they are second only to USB mics in terms of the 'difficult' stakes - search this forum!
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Windowing an OutputSream from the mic .. help !

Hi all ..
first I'd like to thank all people here in the forum .. i learnt a lot from u guys !
My problem is ..
I'm using J2ME
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the code i use , makes the recorder sleeps for the 3 seconds .. then all data time samples are available at the end ..
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Sorry, wrong thread......
Here is some info about the applet tag
http://java.sun.com/docs/books/tutorial/applet/appletsonly/appletTag.html
Allthouhg that one still uses CODE="MyClass.class"
and should be CODE="Animator" (without the .class)
The above link tells you about the properties of the applet tag but you should use the object tag instead:
http://java.sun.com/j2se/1.5.0/docs/guide/plugin/developer_guide/using_tags.html
Allthough that article mentioned situations where to use the applet tag.
The only reason to use applet tag is if you want the applet to be run with MSJVM.
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AA3 Sound input from computer mic port-help needed

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