Optimizing microphone input level into GB with a mixer

I am trying to optimize the mixer settings for recording from a professional quality studio condenser microphone (requiring phantom power) in GB. I have the options of 1) changing the preamp gain on the microphone input, 20 changing the level of the microphone channel in the main mix, and 30 changing the main mix output level. These output from the mix goes through an iMic into a USB port on my mac.
Can someone describe how to set the mixer output to maximize the signal-to-noise ratio without causing distortion in GB? When I have followed the general recommendations for setting the gain, level, and main mix output for live performances, the output from the mixer causes GB to distort, even though there is no clipping occurring before it leaves the mixer. I can turn down any of these settings and get a decent recording in GB, but I'd like to know how to optimize the settings and not just settle for something that just works.

Consider an all-digital chain of signals (for sake of the argument, in analog terms it's similar), recording at 16 bit. If at the first stage of your recording you use only 12 of the 16 bits, you're giving up half of your resolution which is important for the dynamic range that your recording has. And this information is lost forever, even if you push the signal at the next stage - you lower the sound quality.
That's why I said that taking advantage of the full bit resolution at every step of your signal chain is crucial.

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How can I set the microphone input level?

How can I set the microphone input level?

It would be best to use the standard windows mixer "sndvol32.exe" (in the windows directory) and control it with the System Exec.vi from the Communication palette.
You may want to look at the following discussion thread, too:
how to control the microphone volume input via LabView
How Can I Record Sound in LabVIEW from Sources other than the Microphone?
Zvezdana S.

Get Microphone input level (amplitude) live

Hi
I'm working on a VoIP application, which has big problems with its echo... (mic->otherusers speakers->his mic->my speakers... i thing you know this szenario...)
I have now the idea to block my partners microphone while i'm speaking, and vice-versa - first of all i know thats not the best idea- since i know that on skype or other VOIP applications echo cancellation works quite well...
but i don' have the know-how how to do that with JAVA-Sound-API (targetdataline and sourcedatalines...)
the problem is, that the program needs to recognize if a person is speaking (to block the other's mic)... i found a VolumeRMS code in this forum (but it seems only to work fine with 8 Bit audio data - and i'm using 16 for better speech quality - the source for this was:
public double volumeRMS(double[] raw) {
double sum = 0d;
if (raw.length==0) {
return sum;
} else {
for (int ii=0; ii<raw.length; ii++) {
sum += raw[ii];
double average = sum/raw.length;
double[] meanSquare = new double[raw.length];
double sumMeanSquare = 0d;
for (int ii=0; ii<raw.length; ii++) {
sumMeanSquare += Math.pow(raw[ii]-average,2d);
meanSquare[ii] = sumMeanSquare;
double averageMeanSquare = sumMeanSquare/raw.length;
double rootMeanSquare = Math.pow(averageMeanSquare,0.5d);
return rootMeanSquare;
There are (i think so) one bug in this code:
its maximum amplitude is at about 0-60 (not 100)
the question is now, how can the program know when someone is speaking? (next to all the background-noise in my room)..
like a "isSomeBodySpeaking(soundbuffer[])" method?
or are there any better ideas to deal with echo-cancellation?
Can anybody help?

user10491813 wrote:
the question is now, how can the program know when someone is speaking? (next to all the background-noise in my room)..
like a "isSomeBodySpeaking(soundbuffer[])" method?You'd have to record the average volume when they aren't speaking and then consider them to be speaking if the average captured volume in a sliding window goes above that threshold...
or are there any better ideas to deal with echo-cancellation?If you're not to the level where you can code a tradional echo-cancellation algorithm, then I'd recommend you simply put in a button the user has to hold down in order to speak...and disable the rendering to the speakers while the user has the button pressed.

Garage Band Input Level

I have not been able to find any control for the input level when recording with garage band on the iPad. I'm working around it, but it's a problem. Is there a control I don't know about?

TexasPengr wrote:
The signal level input registers on system preferences>sound>input source, but no matter what I do, I can't get Garage Band to recognize a signal input.
set it in GB:
http://www.bulletsandbones.com/GB/GBFAQ.html#selectinterface
(Let the page FULLY load. The link to your answer is at the top of your screen)

Setting Input Level with Griffin iMic

I'm trying to digitize an audio cassette with Soundtrack Pro. My setup includes a Griffin iMic that goes to a USB port. The trouble I'm having is in controlling the input volume. The level meters indicate clipping, but I'm not clear how to set the input level.
If it's in the Recording tab (>Input) with the Gain slider, mine is greyed out.

I'm trying to digitize an audio cassette with
Soundtrack Pro. My setup includes a Griffin iMic
that goes to a USB port. The trouble I'm having is
in controlling the input volume. The level meters
indicate clipping, but I'm not clear how to set the
input level.
If it's in the Recording tab (>Input) with the Gain
slider, mine is greyed out.
M Wohlgy,
(I know this is months after your post.)
Control of the iMic volume is in the Preferences pane called "Sound". You can set the output and the input levels there.
Remember the iMic's position switch works only for the imput, either close to the microphone icon for "microphone" or away from it for "line in".
I use the iMic exclusively.
David

Multi-track Recording with Audigy (input level adjustment of 6 tracks)

Hi,
Has anyone used an Audigy (or similar) to record more than 2 channels?
I have an Audigy 4, which I use for 24 bit recordings. Usually I just use 2 channels, and an ASIO4ALL v2 interface which probably uses the "WDM:SB Audigy 4 Audio [D880]" driver.
However, there's another driver I can select which is "SB Audigy 4 ASIO 24/96 [D880]", and this gives me 6 input channels corresponding to the R and L channels of the inputs line , 2 and 3. This is great, and seems to work OK, but whereas I can control the levels of the incoming signals (on line 3 L/R) with the 2 channel driver, the Creative surround mixer doesn't have any control over the input levels in the 6 channel case, so I'm stuck with having to adjust the level of the incoming signal externally.
Has anyone found a solution to this difficulty? I've updated to the latest drivers.
I'm using Windows XP pro. and the recording software is n-Track 5.1.
Thanks,
Peter

Marskat wrote:
HI, I dont know if anyone has had this happen to them or not. I Have a MODEL # SB0090 Creative Labs Sound Blaster Audigy with 394 PCI Sound Card with the front controls. I have used it for years. I just re-loaded it into a new computer and loaded the new driver. When I had the old driver loaded I got a very harsh repeated ......BLAP.....BLAP.......BLAP.......BLAP it was very nasty and sounded like a cd skipping. I loaded the new driver and that got rid of the BLAP skipping thing. Now with the new driver when I bring up the windows play and record controls and when I have the Mutes all off and say for example AUX 2 selected, as soon as I arm the track in Sonar the AUX 2 on the windows play control auto mutes right before my eyes I can then de-select mute and the track stays armed, BUT as soon as I hit record it will auto mute again yet the meters still move and it still records and I am able to de-select mute again while recording and it does not effect the recording. It is a very big hassle during a recording project. ANY IDEAS WOULD BE AWESOME!!!!
Are you using ASIO (or asio4all), WDM/KS or MME32 on Sonar?
Not sure but have you checked how the "Record Without Monitoring" (or similiarly named ) option is set (Start menu, Run..., sndvol32 /r).
On Sonar you may need to enable Input ECHO ( [color="#339966"][)))]? button on track headers).
jutapa

How to input data into a table with columns?

I am trying to input data into a table. My table have 5 columns and an unlimited amount of rows. I created a program in LabView that enters the data into the table but it enters all of the data in one row. I would like to enter the first set of information into the first column, the second set of info into the second column and so on. I am including a copy of the program that I am working with. I would like the number of runs to be put into the first column (it should count down like number 5 in first row, number 4 in second row, number 3 in third row, and so on). I would like the applied voltage to be placed in the second column, and so on. Any help or information will be greatly appreciated. I am working with LabView Version 6.1 and 8.0. I am submitting the vi from 6.1.
Attachments:
FJ-PROGRAM.vi ‏68 KB

Pondered,
I looked at your code and I think you might be making things too complicated. I've included a very simple example that demonstrates how to write a 2D array of integers to a table. Hope you find this helpful. It is in LV 7.1.
Chris C
Chris Cilino
National Instruments
LabVIEW Product Marketing Manager
Certified LabVIEW Architect
Attachments:
rows - columns.vi ‏17 KB

Is it possible to input itunes into DVI/HDCP input from imac with Apple mini display portS?

Is it possible to input itunes into DVI/HDCP input from imac with Apple mini display port? Or is there another way?

The problem with itunes is needing a computer running it. or continuous internet connection.
A Time Capsule is AC mains powered.. no battery. Although you can certainly buy an older Gen4 TC and modify it ...
https://sites.google.com/site/lapastenague/time-capsule-power-supply-repair-kits
I have run my TC on a 12V SLA battery.. which would neatly meet that part of the functionality you want. But it is heavy and not really the rugged sort of device I think you want.
But I would look at other media players and other ways of storing movies than iTunes.. if you want to carry the whole lot around with you .. A bunch of SD cards even you could plug into the ipad??
External storage for iPad?
A small cheap PC/Mac laptop with a 1TB hard disk in it.. and 11" or 13" screen.. can often be a much better solution than lugging around multiple items so you can fit Apple devices into your life.

Can't adjust input level when using usb microphone

Hey guys,
I'm very familiar with this version of Garageband, and just started using a Blue Snowball USB mic. When I plug it in and set the drivers accordingly, the input volume slider is greyed out, making it inaccessible. Is there a trick to being able to adjust input volume?

Option Click on the Volume slider icon in the menu bar and choose Sound Preferences. Choose the interface and you should be able to see the input level.

I'm trying to input phone numbers starting with 0, into Numbers spreadsheet. It keeps deleting the 0- any clues?

I'm trying to input phone numbers starting with 0 into Numbers Spreadsheets. It keeps deleting the 0. any clues anyone?

select the cells (or column) where you are entering the phone numbers and change the format to "Text":
Then "Natural" way to write numbers is to omit leading zeros.
Another ways is to leave the the cell format as "Automatic" and type a single quote BEFORE entering the phone number... like this:
'0408

I am not getting any sound input on my microphone. There is no microphone input on my new macbook pro, so I am using my old one. Is this a problem with the new operating system? Did they simply remove any microphone software?

I am not getting any sound input on my microphone. There is no microphone input on my new macbook pro, so I am using my old one. Is this a problem with the new operating system? Did they simply remove any microphone software?

Check System Preferences > Sound > Output
Make sure the correct device is selected.

I connected my iPod Touch to a stereo system phono input using a cable with stereo jacks (red and white) on one end and that plugs into the iPod's headphone jack. The problem is that the sound is extremely muffled and bass-heavy, with little treble.

I connected my iPod Touch to a stereo system phono input using a cable with stereo jacks (red and white) on one end and that plugs into the iPod's headphone jack. The problem is that the sound is extremely muffled and bass-heavy, with little treble. I have to turn the bass control on the stereo receiver to pratically zero in order for the sound coming through the stereo to be acceptable. Is there a setting on either the iPod or the stereo that I need to change in order to fix this?

The phono input is for a phonograph. The iPod does headphne output is not that same as a phongraph output. Does th stero have an AUX input? That should be compatible.

How to Set Input Levels with FirePod and GarageBand

Hello ....
I/ve got Garageband and my Firepod 8 channel interface working ... but I need some tips .. especially on the relationship of input levels between my Firepod and Garageband ....
When I am setting input levels for a recording ... where should the input level be on the track in Garageband? .... Is that volume setting only for playback and mixing? .... or is it for the input as well ...
Thank you ....

Where should the initial input levels be set? ... Would you use the loudest
part of the song to do this? ...
Yes, as loud as possible without lighting the FP's clip leds.
When I mix the final product ... and change the volumes on the different
tracks.. aren't you then changing the input levels?
No, you're changing the output levels.
can't you red line and overload the track?
Yes, that's why each track has a fader, so that you don't clip.
Is there a master ouput level indicator
Yes, right over the Master Volume slider.
I guess you can't just put each track as loud as you want ... correct?
As loud as you want without clipping.
I would strongly suggest doing some web searches on "Mixing and Mastering" to learn more about how to mix, there's a TON of info floating around the web.

How do I connect an external microphone to a Macbook Pro with Retina display ?

How do I connect an external microphone to a Macbook Pro with Retina display ?
Alternatively how do I connect an audio input at a level that would normally have gone into a 'line input' socket ?

Plug it into the linein/headphone jack if there's just one. It'll switch between the two modes automatically depending on what you plug into it.

Input data into dynamic calc in Planning forms

Hi guys,
Have someone tried to combine these two requirements:
1. to have upper level members dynamic calc
2. to give users the ability to input data into those upper level members through Planning forms.
I need to do that in order to pass inputted values into HBR, and allocate those values to the leaf level across multiple dimensions. I need those members dynamic calc because otherwise I would need to aggregate data across 7 dimensions and that takes too long. Those allocations are suposed to happen constantly and are the main purpose of application.
I know you can have dummy stored members that correspond the uper level dynamic members, but then the user will have input forms without current data and will be required to check reports/forms with dynamic calc members prior to updating data.
I wonder if it is a realistic task to change jsp files that generate those forms, so that data can be inputed into the form, but instead of passing it to Essbase cube to send it to relational database.
P.S.
This is related to the thread I opened about optimizing recalculation time:
Recalculatiion takes much longer than initial calculation
I received a lot of valuable advices that would definitely work in other situations, but in my case they seem to contradict either user concurrent access or performance requirements.
Thanks a lot!
Dmitry

Hi Glenn,
I think i found a workaround for this, or may be i am missing something. I will replicate my hierarchies so that one hierarchy will have uper level stored members, and another dynamic calc. Once user loads the form it will run on-load script that will copy data from corresponding dynamic calc to stored slice. The updated data will be alocated to the leaf level, but will never be aggregated. Since reporting is done from ASO cube i only care about the leaf level and current selection in the form.
Thanks!

Optimizing microphone input level into GB with a mixer

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