OS X - Multiple Microphone Input

I find that when I us Microphone.get and attach() a
microphone to a netstream, it works fine.
When I attach a second microphone to record a second stream,
it works fine on Windows but in OS X it fails, the stream just
records an empty zero-length file, the input-level stays constant,
never changing.
Is this a known problem? Is there a work-around of any
kind?

I find it staggering that Creative produces cards that have so many problems!
I realise that they are popular (over-promoted?) products and this forum is likely to be populated with users having a problem, but the sheer range of problems with the SB series of hardware and examples of the poor support Creative gives (eg drivers that don't work with dual core/ multiprocessors until recently, many reported difficulties with multiple sound sources) is extraordinary.
Let's face it, most people buying these cards are doing so to run Teamspeak/Ventrillo plus a game. Any soundcard that can't easily cope with this is utterly useless.
For me, it'll be try the helpline tonight and RMA if they can't sort it.
Creative, you are useless!
phoenix963

Similar Messages

Usb 6009 multiple analog inputs

I am currently attempting to sample two different analog inputs at different sampling rates using a USB 6009. I keep getting the 'resource reserved' error and am wondering if this is not possible using this DAQ. Questions:
1. Does creating two analog input channels on the device cause this error?
2. Is it possible to sample at different rates on channels created in the same task? (i am trying to 'slow down' the second analog input to display switch points to a customer)
3. Running multiple analog inputs using independent timing would be better achieved by switching to a higher end DAQ? If so which would you recommend?
I have attached my vi. Thank you in advance for your help.
I surf therefore I am....
Attachments:
demo_nolvl.vi ‏27 KB

The DAQ boards only have 1 timing clock for the analog inputs/outputs. So you can only have 1 sample rate on a given card. I would recommend just sampling at the highest of the desired rates in a single task.
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines

I want to be able to use airplay to stream audio to another iOS device, and then use that audio stream to be used when recording video instead of the built in microphone / microphone input. Is this possible?

I want to be able to use airplay to stream audio to another iOS device, and then use that audio stream to be used when recording video instead of the built in microphone / microphone input. Is this possible?

A third-party app probably cannot obtain a stream from another app. To the best of my knowledge, such a capability is not provided in the software development kit, apps being "sandboxed" from each other and so allowed to communicate only in very specific and limited ways.
I'm not completely sure what you mean by "limitations on video capture". An iPhone, to the best of my knowledge, can natively record video only through it's built-in camera, and audio while doing video recording only through the built-in microphone or mic/headphone jack. There might be a video recording app that would allow audio input from an external device connected to the dock connector, but I'm not sure.
Regards.

Synchronizing two counter frequency inputs with multiple analog inputs

Hello all,
I'm fairly new to LabVIEW and I'm trying to collect data from multiple sources with synchronized timing on the acquisition but I'm having trouble figuring it out. My problem is that I've got two counter frequency inputs, one optical tachometer reading one pulse per revolution, and a max machinery flow meter with a k factor of 12000. I can't seem to figure out how to sync the timing with my multiple analog inputs. I've been attempting to get the tachometer to sync with the analog inputs first by following the example linked here. (https://decibel.ni.com/content/docs/DOC-10785) So far each time I run it I either get a timeout error on the DAQmx read or a "Multiple sample clock pulses were detected" error (see attached image). It seems if I slow the sampling rate way down to say 10 hz and ensure that the tachometer signal is over 800-1000 RPM (13-17 Hz) before starting the VI then the program will run without errors until the RPM drops below that threshold then the "Multiple sample clock pulses" error occurs. The code is attached below.
Does anyone know of a more effective way of syncing counter frequency inputs with analog inputs? I'd like to have a VI that can show 0 RPM (and eventually 0 flow as well, but I think I need to figure out the timing of one counter before I add another as it seems I can't have two counters in the same task). Any help on this would be greatly appreciated.
LabVIEW version 13.0
cDAQ-9178 Chassis with NI 9401 for the two counter inputs and NI 9205 for the analog inputs.
Thanks!
Richard
Solved!
Go to Solution.
Attachments:
SimpleDAQ.vi ‏44 KB
LV_Error.JPG ‏31 KB

Maybe third times the charm?
So I've finally got a good handle on why the VI is having problems at low RPM though I'm somewhat embarassed how long it took me to do that
Because I have the counter time synced to my Analog input task if it doesn't see at least two pulses between the two clock pulses set by the analog input task I get the -201314 "Multiple sample clock pulses" error. This seems fine at first as it just sets a minimum RPM that I can measure and it's well below the area I'm interested in so no problems there. I tried a simple error handler that would clear the error when it happend assuming the loop would keep iterating until the RPM went above that minimum at which point I would get a signal again. This is not the case, the read function just continues to spit out the -201314 error even after the RPM is back in the readable range. So then I tried adding two case structures so that when the error occured it would stop the task, clear the error, and then start the task again on the next loop iteration (Code Attached). This also doesn't work as the error shows up again on the stop task and then AGAIN on the start task on the next loop iteration. It seems this error is not actually being cleared and once it happens it stays with the task regardless of what the error cluster is carrying.
Anyone have any ideas? The only solution I can think of is to just clear all tasks and recreate them each loop iteration until the RPM is readable again but that strikes me as a horribly clunky solution.
Richard
Attachments:
SimpleDAQ_1_Start Stop.vi ‏48 KB

External Microphone Input?

The iPhone headset's (or similar one plug headset) microphone and headphone can be used by plugging the headset into the headphone jack. In Mac OS X, the internal microphone option changes to external microphone.
Boot Camp doesn't seem to recognize this. All I get is the internal microphone, the line-in jack, and the digital input (SPDIF).
Is there a way or a driver to enable the use of the microphone input for the headphone jack?
Thanks!

first of all, i don't work for lenovo.
i'll try to find a specific answer for you. ( i will post here if i can find )

Multiple MIDI Inputs Yet???

Can anyone tell me if LP8 offers multiple MIDI inputs? I use the step sequencer from my Elektron Monomachine for pattern creation and would like to be able to assign the external tracks from it to instrument tracks hosting either native or 3rd party plug-ins. Untill I can assign MIDI to Logic Instruments and third party AU Instruments from multiple channels, I am forced to use another app (Ableton Live at present) for song creation and only use logic for post production of said track. Would much rather keep it all in one app from creation to stage performance!!!

I have found an answer to my problem. In the following link I found this post that allowed me to play multiple instruments from an external step sequencer. I have been trying to figure this one out for ever and just kept giving up with the environment. Now I can finally use logic the way I always wanted to!!!
http://discussions.apple.com/thread.jspa?threadID=1078171&tstart=30
Here is the post...replace "MIDI Keyboard" with "Sequencer Track" and you can see how this worked for me...
This method assumes that One MIDI Keyboard is sending on MIDI channel 1 and the other is sending on MIDI channel 2.
In the Environment, go to the "Clicks and Ports" Layer. My MIDI "signal" path is:
Physical Input-->Input Note (a keyboard)-->Input View (Monitor Object)-->
"to Recording and Thru"
(See p. 207 of the LPReference for more on the "The MIDI Signal Path")
What I would suggest is this:
Create a "Channel Splitter" object from the "New" menu. Then cable the MIDI output of the Input Note (the Keyboard object) to the Channel Splitter. This is a second MIDI output cable from the Keyboard, make sure the keyboard object remains connected to the Input View (monitor object).
Now create two "Alias" objects. In the Alias' Parameters box, set the "Reference" box to the Audio Instrument you want to hear when you play the MIDI channel 1 keyboard. (There are other ways to do this without alias objects as well.)
Now cable the MIDI channel 1 output of the Channel Splitter (second from the top) to the Alias of Audio Instrument (the one to be heard when playing the Keyboard on MIDI channel 1).
Create a second Alias, set its Reference to the second Audio Instrument and cable the Channel Splitter's MIDI channel 2 output to the Alias.
Now go the Arrange. Create a new track and Assign its output (click-hold on the name of the track) and select the "Input Notes" object (mine was under the "Clicks and Ports" submenu). Now when you've got the Input Notes track selected in the Arrange, the MIDI notes will be sent to the Audio Instruments based on MIDI Channel. You can record the notes on the Input Notes track. Later, you can "Demix" the notes and put the Demixed regions onto the tracks of the appropriate Audio Instrument.
End Post
For now I am using only one port on my AMT8 for input, the next step will be to try multiple port / multiple sequencer input scheme and see how that setup works...
Hope this is helpful to someone

Never used microphone input before, sound recorder window okay, no sound output.

Sound from sound recorder input goes nowhere, a sound window is supposed to open, it does not.
There is some software missing in my pavilion g7 laptop. Media player sound is fine, internet sound
is fine, only sound missing is from the microphone input. The sound recorder window is open and the
recording bar is moving as the input varies, how do I get the sound to the speakers. Never used this
option before, it might be defective from the factory.
This question was solved.
View Solution.

Problem solved. You cannot turn on computer with the microphone or other audio cable plugged in.
Otherwise the computer does not respond to the inputs, you have to plug in the microphone or audio
cable after logging in and then a window will open that allows you to select the that input over the default.

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.

Best way to clear multiple textbox inputs?

In an app with multiple textbox inputs, is there a way to iterate through all controls which are textboxes in order to clear their text inputs by using a loop? I could write multiple "mytextbox.clear()" statements, but I was wondering if there is a better way.
That is, as a more general question, does JavaFX have a way to loop through similar controls so that batch functions can be performed on them?

I was speaking about these methods without keeping a reference to the Textboxs, like in full declarative syntax.
Stage {
    def tbIds = for(n in [0..<10]) "TextBox{n}";
    var vbox: VBox;
    scene: Scene {
        width: 100 height: 250
        content: [
            vbox = VBox {
                layoutX: 10
                layoutY: 10
                content: [
                    HBox {
                        content: [
                            Button {
                                text: "Set"
                                action: function() {
                                    for (id in tbIds) {
                                        //def tb = vbox.lookup(id) as TextBox;
                                        //Also work
                                        def tb = vbox.scene.lookup(id) as TextBox;
                                        tb.text = id
                            Button {
                                text: "Clear"
                                action: function() {
                                    for (id in tbIds) {
                                        def tb = vbox.lookup(id) as TextBox;
                                        tb.text = ""
                    for (id in tbIds) TextBox {id: id}
}

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.

Time synchronization problem with "niUSRP EX Rx Multiple Synchronized Inputs.v i " ??

Hello,
I used "niUSRP EX Rx Multiple Synchronized Inputs.vi "( offred by NI) to synchronize 2 USRP in reception (the master connected to laptop via Ethernet and the slave connected to the master via MIMO cable).
I set: master--> RefIn and PpsIn     slave-->Mimo and Mimo.
Problem: Inspite I used identical cables that connect the both devices to the transmitter; the signal received by the master and the signal received by the slave ar'nt synchronized in time because the time lag (delay) is not constant!!!.
Where is the problem?!!
Thank you

Hi,
I used:
transmitter--> Rohde & Shwartz SM300 signal generator.
Receivers--> 2 x USRP N200 (connected with MIMO cable) master--> RefIn and PpsIn     slave-->Mimo and Mimo.
3 identical cables to transmit the signal, connected via Power Splitter/Combiner (Mini-Circuits).
Cable 10MHz connects the master device with the transmitter.
FreqTx=700MHz (sin)           CarrierFreqRx=701MHz --------> so the received signals are a sin of 1MHz (701-700=1MHz).
I have no probleme with phase synchronisation (because both the slave and the master received a sin of 1MHz).
My problem is the time synchronization:
How can I verify if i did the time synchronization?
I must have 2 superposed sin [(ch0 I with ch1 I) (ch0 Q with ch1Q)] ?
I must do Timed commands? if yes! How to do it?
I try to run the example of constellation, then i have message: Find VI Named".........Vi"    I have this problem all the time when I run some examples . I have probleme with setup LabView?!
Best regards

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.

Can I use a MacBook's Only headset port as a Microphone INPUT?

Hello Apple Community,
Can I use the Macbook Pro's only one headphone port on the left side of the machine for a Microphone INPUT as well? Is there a setting in System Preferences so that I can change it? I can't find anything.
Thank you for the help!

Hi Needa_Pickle
Unfortunately there is no native setting to do this and some third party applications can cause the system to "forget" the port and leave it redundant. It can be done however, I bootcamp into windows and work on that side due to the limitations on the mac, I find this a shame as its one of the few problems I have the mac. Hope this helps

Constantly record and analyze microphone input

Is it possible to analyze microphone input data while it is being recorded? My program needs to measure the input voltage and the program needs to procede when the voltage is equal to 5V.
Thank you.

The sound card can be a bit problematic in this sense especially if you change the input gain for the mic and watch out for any bass boost function being on used as this will affect the scale as well.
As for the scaling on a standard sound card, there is none as such (see article 3 below). The input
is represented as ±32767 on a standard 16 bit card and obviously larger
for the newer 24bit cards.
The easiest option would be to generate a signal, measure it and then play into the mic input. I would suggest that the mic input (or possibly better the line input) would be in the order of a couple of hundred milli volts so if youre sure that you have a five volt signal then you will need to scale the input with a resistor (say a 5K ohm variable resistor set up as a potential divider).
You may find that the following articles will help not only with the principles but with the practice as well: -
http://www.virtins.com/Virtins_Sound_Card_Oscilloscope_Probe_Manual.pdf
http://www.techmind.org/audio/
This link is quite good as well:-
http://www.audioprecision.com/index.php?page=support&id=1100001014
If its a mac the following may apply (I don;t use them a great deal, sorry): -
http://www.channld.com/distspec.html

Are iTouch 1st gen users blocked by Apple from Microphone input:

I have wanted to get a microphone to use my trusty iTouch 1st gen for voice memos. Since the 1st gen never came with a mic in the plug spot, McAlly and a few others made an adapter to input through the dock connector.
However Apple quickly changed the firmware from 2.0 to 3.0 and blocked this feature. Is there a work around for this issue. My 1st gen is a reliable unit and I would like to not have to buy another.
Thanks, GB

answer is they don't Gen1 iTouch never had microphone inputs.

OS X - Multiple Microphone Input

Similar Messages

Maybe you are looking for