Sound Input - Help

G'day,
I'm trying to record some audio into my eMac 1.25Ghz...
Am I right in thinking that the Audio Line In port on the eMac requires a powered source, and thus an unpowered microphone hooked up to the 3.5mm audio jack will do - nothing?
I'm trying to use just a plain old 1990's Apple microphone (also tried some headphones... seemed to work in the "old days") but get no levels...
Also - my eMac's sound input and output options have selections for "Soundflower" 2ch and 16ch.... What on earth is that? :} I'm assuming it's from something I've installed in the past...?
cheers
cosmic

Well, for future queries on this topic...
The eMac audio input jack is LINE LEVEL... Thus you need a powered device to gain any kind of signal on it. In other words, a plain normal microphone will not work.
In my case, I found a dicta-phone (sony cassette based voice recorder), and plugged it in via a mini-din to mini-din cable... (via it's headphone socket). Headphones are powered, thus it gave enough power to obtain a signal.
cheers

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Either remove "Soundflower" or refer to its developer for support.
Any third-party software that doesn't install by drag-and-drop into the Applications folder, and uninstall by drag-and-drop to the Trash, is a system modification.
Whenever you remove system modifications, they must be removed completely, and the only way to do that is to use the uninstallation tool, if any, provided by the developers, or to follow their instructions. If the software has been incompletely removed, you may have to re-download or even reinstall it in order to finish the job.
I never install system modifications myself, and except as stated in this comment, I don't know how to uninstall them. You'll have to do your own research to find that information.
Here are some general guidelines to get you started. Suppose you want to remove something called “BrickMyMac” (a hypothetical example.) First, consult the product's Help menu, if there is one, for instructions. Finding none there, look on the developer's website, say www.brickmymac.com. (That may not be the actual name of the site; if necessary, search the Web for the product name.) If you don’t find anything on the website or in your search, contact the developer. While you're waiting for a response, download BrickMyMac.dmg and open it. There may be an application in there such as “Uninstall BrickMyMac.” If not, open “BrickMyMac.pkg” and look for an Uninstall button. The uninstaller might also be accessed by clicking the Customize button, if there is one.
Back up all data before making any changes.
You will generally have to restart the computer in order to complete an uninstallation. Until you do that, there may be no effect, or unpredictable effects.
If you can’t remove software in any other way, you’ll have to erase and install OS X. Never install any third-party software unless you're sure you know how to uninstall it; otherwise you may create problems that are very hard to solve.
Trying to remove complex system modifications by hunting for files by name often will not work and may make the problem worse. The same goes for "utilities" such as "AppCleaner" and the like that purport to remove software.

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Thanks.
G4 desktop Mac OS X (10.4.6)

Hi, Neil -
This Apple KBase article provides some info with regard to microphone selection -
http://docs.info.apple.com/article.html?artnum=18275
Scroll down the article to the G4 section that matches your G4 model.
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Edited by: Tiberiu Sasu on Apr 27, 2010 11:25 PM

Just throwing ideas around...
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Is this possible in iOs.

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!
A much, much better way to attempt to do this would be to use a sensible ASIO external device with sufficient inputs, and a mic that plugs into it, not a USB one. That way you could route the inputs and outputs appropriately, and use the device's monitoring so that you could monitor what you need to. I wouldn't attempt to do anything like this any other way, I'm afraid. Another approach (which can be made to work well) is to use an external mixer to do most of the work - often it's easier to set up the monitoring on these. You can get mixers with the appropriate interface built-in, and they don't cost a fortune either. Mackie and Behringer make suitable devices.

Alternative sound input

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Welcome to the Apple Discussions!
Since folks have been using GarageBand for a number of years successfully with their mics and instruments, perhaps it is the operator...
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But if your kit is producing line level input, perhaps you have not configured GB correctly for recording. In the GB Help menu you will see access to a getting started with GB pdf. In Chapter 5 it explains how to configure GB for voice recording.
Let us know if nothing I have pointed to solves the problem.
Dah•veed

Sound Input Read VI hangs

I'm using the SI Read VI to sample from the Soundblaster card in my Dell PC. I've configured the device for 16-bit mono input, sample rate 44.1 kHz, and the buffer size is 1024 samples. The SI Read VI is in a Timed Loop whose period I've set to 23 ms (1024 / 44.1kHz). Following each acquisition, I do some filtering, frequency analysis, and plotting inside the loop.
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This message is a reply to "Sound Input Read VI hangs" (National Instrument) NI Discussion Forums posting found at: http://forums.ni.com/ni/board/message?board.id=170&message.id=137258 and http://forums.ni.com/ni/board/message?board.id=170&message.id=110656&requireLogin=False
Hi everybody,
From the large count of how many of you guys read the above thread of messages, including myself, and who are dealing with the same problem of SI Read VI hanging found National Instrument (NI) LabVIEW (LV), I decided to share my (not so perfect) solution with you guys to save you time and frustration. Here is the story of 3 days (18 hrs/day) of my life, fighting a bug in SI Read VI (SI: Sound Input) (I am currently using LabVIEW Pro v 7.1.1):
For the reason that "A Message cannot exceed 10,000 characters" on this Forum I included my message in the attached Word doc. Hopefully, there is no such limitation on uploaded files!
Samir Berjawi
Research Assistant and Lab Instructor
American University of Beirut
[email protected]
[email protected]
Attachments:
LabVIEW SI Read freezes - Report.doc ‏39 KB
Sound Acquisition Test.zip ‏995 KB

Digitech RP350 USB Guitar Pedal, Not picking up sound input

I just was trying to hook up my guitar pedal to garageband to record some tracks. I have the MIDI set up right I am pretty sure, garageband even recognizes the input device as "Digitech RP350". But when i try to record, it is picking up no sound input. I just want to get this to work and any help would be greatly appreciated.
Thanks
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...and not to be annoying, but, skaadiction...you get much better control of your signal if you go out the normal 1/4" jack(s) and into GB via an audio interface. Using the trim on an interface will allow your recorded signal to be much hotter/thicker/stronger than by going straight from USB, where it seems you have no control over the incoming signal.
(Just talking from my experience. Someone correct me if I'm wrong.)

Sound Input Read Spectrum Analyzer

I am working on a project with Arduino to do a spectrum analyzer. The LVIFA part of it is working fine.
The part that I am struggling with is the audio input.
I am using the Sound Input Pallet.
My troubles are two-fold.
1. A single read takes quite some time and I need to have a pretty quick response. I decreased the samples per read and came up with problem #2.
2. I can get one read out of the Sound Read In and then the VI locks up. I have my timeout set to 1 Sec and it never times out. I have to restart LabVIEW.
While it would be nice to have this VI working, I'm not even sure that I am using the correct tool. If I am going to be running an audio spectrum analyzer display off of the data input, I need to response to be within a few mS (ie. <200) so that it looks visually correct.
If this library will not achieve this, then I need to look into a different tool.

Hi,
I have a similar project, actually I am using a Audio frequency range as input to the Arduino and do an FFT to receive the spectrum out of it.
See the YOUTUBE video I did and the code in *.jpg. I took actually the Continous Input example and modified it.
Hope that helps.
Keep me posted, since we might work on a same project.
http://youtu.be/aqzyofQHXDM
Y3G
Attachments:
Uno Blockpanel.JPG ‏104 KB
Uno FRONTpanel.JPG ‏99 KB

PLEASE can a AE from NI take a look at my problem. Sound input read behave in strange manner then the buffer size is larger than 2X number of samples to read.

On my computer I have discovered some strange behavior then reading data from the sound card. Then the buffer size is 2x samples to read everything is as expected. But since I read the sound card 10 times pr second I feel a .2 second buffer is to small. I am using XP, and XP is not a RTOS so with a buffer set to 0.2 seconds I may lose data. Therefore I set the buffer size (number samples/ch on Sound Input Configure.vi) to be in range of 2 seconds. The result then is that then reading from Sound input.vi, a reading often take more than 0.1 second. On my computer it is often 500mSec. Then the next 5 read follows with almost zero interval. I do not loose data. But on my front panel the graphs looks like an very early silent movie. This error was introduced in Labview 8.x. To be honest I think the labview 7.x sound system was much better in many ways.
But before I point any finger NI. Other people has to verify the behavior I experience. I have made an example showing this error. It is a modified version of the "Continuous Sound Input.vi" example. Then the "buffer in seconds" control is set to 0.2 every thing works OK. Changer this to a larger number will produce the mentioned above hiccup. The larger number in this control the larger hiccup. Is it any way to fix this? My solution up to now has to use a free 3. part software(http://www.zeitnitz.de/Christian/index.php?sel=waveio) But I guess it soon will be outdated. It may not work with newer windows versions.
Any help at all will be appreciated
And yes I have the most updated version fo DirectX. Also I se this in Labview 2009 which I have trail version of. The VI I have made is in 8.6
Message Edited by Coq Rouge on 09-07-2009 10:54 AM
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)
Attachments:
Continuous Sound Input with timing.vi ‏23 KB

macaba wrote:
If you take a moving average of the 0.2s buffer vs. 3s buffer at an update rate of 10, then they are the same (just under 100ms), so the average refresh rate is the same. I agree that is odd behaviour that the time between sound reads go to zero quite a lot then take a long time once in a while (presumably to fill the buffer
I guess it goes to zero because it is reading data from the buffer it do not has to wait for data from the sound card. The mysterious thing is the periodic delay. You are also correct then saying that average timing is correct. And in my application I have no data loss.
If you search for sound in this forum you will find out that many people has reported trouble with the sound system.
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Sound input plot update problem

Hi,
I am having some problem regarding the plot of the sound input signal. Specifically the problem is regarding the update of the waveform graph i have used.
the update is not continuous mode. And there are gap in the waveform
can anyone help me
Attachments:
Continuous Sound Input.vi ‏20 KB

Thanks! Though I upgraded to solve the problem, this is good to know.
Bob
Date: Wed, 9 Nov 2011 04:45:42 -0700
From: [email protected]
To: [email protected]
Subject: Premiere Pro 2.0 and Windows 7 - Direct Sound Input soundcard driver problem
Re: Premiere Pro 2.0 and Windows 7 - Direct Sound Input soundcard driver problem created by Steve-Italy in Premiere Pro CS4 & Earlier - View the full discussion
VERY VERY VERY EASY (now, when I solved the problem, not before...!!!)
Insert Micropone or any jack in Micropone Input or Aux Input, on your sound card or motherboard !
AND NOW YOU CAN USE PREMIERE PRO 2.0
Enjoy
Steve - Italy
Replies to this message go to everyone subscribed to this thread, not directly to the person who posted the message. To post a reply, either reply to this email or visit the message page: http://forums.adobe.com/message/4015730#4015730
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Time mismatch with Sound Input Read VI

Hi Folk,
I am acquiring the signal of the PC
sound card with the "Sound Input Read VI" and I have
noticed that between subsequent waveform data packets there are time
mismatch, both overlap and gap.
To point out the observed behavior, I
have posted a modified example, the "Continuous Sound Input.vi". In the example, I have computed the
time difference between the t0 of the actual waveform packet and t0
expected on the basis of the previous waveform packet.
Consistently, in the indicator "Time
series" (Waveform Charts), each time there is a time overlap or
gap the Charts resets it self or presents a gap.
By reading 1 second of data the time
mismatch is about 0.015625 or 0.03125 sec. (both positive and
negative).
The repetition frequency of the time
mismatch decreases as the acquisition sample rate increases.
The amount of the time mismatch seems
to be sample rate invariant.
Do you have any idea from where this
problem is coming out and how to solve it?
Thanks for your help,
Asper
Attachments:
Time mismatch with Sound Input Read VI.png ‏51 KB

Pre made Labview functions are not some holly grails that is newer to be touched and modified. In fact many included functions in Labview has what I will name as "high flimflam factor" That will say a lot of functions you do not really need. Express VIs are grim examples of this.
Anyway I have made some modifications and removed some babyfat in the sound input VI. Take a look at it. The top level VI is the "time fixed sound.vi" It could be that you will get an error because Labview will not find a DLL. If a DLL is reported missing you will find it in C:\Program files\National Instruments\LabVIEW 2010\resource\lvsound2.dll
Remember to save the modified VIs in a separate folder, and not in the vi.lib folder at all. Be careful so you do not overwrite any Labview function
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)
Attachments:
Sound Input Read (DBL)_time_fixed.vi ‏30 KB
time fixed sound.vi ‏19 KB

G5 sound input reads drums but no sound coming out

i'm not sure what the deal is, the input is reading my drums fine but i hear no sound not through headphones or surround sound or computers speakers. All my sound works for anyhting else. please help me

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Stft from sound input VI dat to plot spectrogram, whats wrong?

Hi everybody,
I listen continously
via a microphone and sound card to 25ms long pulses in the range of
9kHz to 12kHz. I get 10000 samples from my Sound Input VI and use the
STFT Spectogram to plot a spectrogram(frequency versus time). Using the
default settings or any other window lengths results into the error
message:"...possible reasons: invalid windows length".
What's wrong?
Thank you very much,
Martin

Hey Brian,
I would like to listen to 25ms long pulses in the range of
9kHz up to 14kHz
+by using a microphone and a standard PC sound card
+continuously up to 1.5 hours.
The VI should display a real-time spectrogram(frequency versus time) like in the example VI "Moving STFT PtByPt.vi". It is important that the spectrogram always shows the last 5 minutes and the current frequencies. It is not necessary that the spectrogram is very real-timed(up to 600ms delay is ok), BUT it has to be 100% CONTINUOUSLY to not miss any of the short 25ms long pulses.
In addition I also would like to determine the frequencies of the received pulses, which are somewhere between 9kHz and 14kHz and I need to count the amount of received pulses.
My issues are:
1. missing some 25ms pulses due to problems with continuous sampling of SI Read VI in a while loop(I guess there is a small break between the 1000 samplesets)
2. How connect SI Read VI output to STFT VI or STFT PtByPt VI to perfom moving spectrogram?
3. Why doesn't show my spectogram no frequenciues at all?
I hope this helps to understand what I try to achieve and in what problems I have run.
Here is a corrected and more use-friendlier version of my VI.
Thank you very much,
Martin
Attachments:
spectrogram_NI.vi ‏142 KB

Sound Input - Help

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