Continous sound output with variable frequency?

Hi all!
I wanted to output a signal via the soundcard with one variable frequency.
How can I do that?
Thanks
ANDY

Hi Andy
I'm afraid I've only got access to 7.0 & 7.1.
SO Set Num Buffers.vi uses a Call Library Function node to access lvsound.dll.
Hmmm...
I've had a scan around and you could try checking this link
It may give you some clues.
Good luck
Neil

Similar Messages

How to structure the DMA buffer for PXie 6341 DAQ card for analog output with different frequencies on each channel

I'm using the MHDDK for analog out/in with the PXIe 6341 DAQ card.
The examples, e.g. aoex5, show a single Timer (outTimerHelper::loadUI method), but the example shows DMA data loaded with the same vector size.
There is a comment in the outTimerHelper:rogramUpdateCount call which implies that different buffer sizes per channel can be used.
(the comment is: Switching between different buffer sizes will not be used)
Does anyone know what the format of the DMA buffer should be for data for multiple channels with different frequencies ?
For example, say we want a0 with a 1Khz Sine wave and a1 with a 1.5Khz sine wave. What does the DMA buffer look like ?
With the same frequency for each channel, the data is interleaved, e.g. (ao0#0, ao1#0; ao0#1, ao1#1, ...), but when the frequencies for each channel is different, what does the buffer look like ?

Hello Kenstern,
The data is always interleaved because each card only has a single timing engine for each subsystem.
For AO you must specify the number of samples that AO will output. You also specify the number of channels. Because there is only one timing engine for AO, each AO will channel will get updated at the same time tick of the update clock. The data will be arranged interleaved exactly as the example shows because each AO channel needs data to output at each tick of the update clock. The data itself can change based on the frequency you want to output.
kenstern wrote:
For example, say we want a0 with a 1Khz Sine wave and a1 with a 1.5Khz sine wave. What does the DMA buffer look like ?
With the same frequency for each channel, the data is interleaved, e.g. (ao0#0, ao1#0; ao0#1, ao1#1, ...), but when the frequencies for each channel is different, what does the buffer look like ?
In your example, you need to come up with an update rate that works for both waveforms (1 KHz and 1.5 KHz sine waves). To get a good representation of a sine wave, you need to update more than 10x as fast as your fastest frequency...I would recommend 100x if possible.
Update Frequency: 150 KHz
Channels: 2
Then you create buffers that include full cycles of each waveform you want to output based on the update frequency. These buffers must also be the same size.
Buffer 1: Contains data for the 1 KHz sine wave, 300 points, 2 sine wave cycles
Buffer 2: Contains data for the 1.5 KHz sine wave, 300 points, 3 sine wave cycles
You then interleave them as before. When the data is run through the ADC, they are outputting different sine waves even though the AO channels are updating at the same rate.

No Sound Output with Digital Output Mac

Hi There,
I normally use my Blackmagic card for sound output on my Mac with Premiere CS6. I want to work with the digital output on my current project but when I try switching to digital out under Preferences>Audio Hardware the audio still comes out of the Blackmagic card but not the digital audio out on the Mac. I'm testing with the bars and tone in the browser and my timeline sequence. FCP 7 works as expected when digital out is selected, and I know digital out is working properly on my computer.
Any suggestions?
Thanks!
Gerry

OK, sorry it was that I had to switch audio device under plackback.
Perhaps this should be added to the FAQ under No Sound During Playback, which doesn't include the Preferences>Playback as part of the troubleshooting for MAC.
Thanks!
Gerry

How to continuously generate a pulse train with variable frequency

Dear all,
I am trying to use NI-USB 6259(BNC) to send Pulse signals to conduct position control of a servo motor with labview. The position control of servo motor follows these rules:
The number of the pulse train determines how many degrees the motor rotate ;( e.g. the angular position of motor)
The frequency of pulses determines how fast the motor rotate; (e.g. the speed of motor rotating)
The digital determines the direction of rotation of motor (e.g. clockwise or counterclockwise)
My question comes when I need to continuously generate a finite plus train signal in a period of time. Below is a sample:
Time(s)
Number of pulses
Direction of rotation
(1 is clockwise, 0 is counterclockwise)
Frequency
0-1
923
1
923hz
1-2
3540
0
3540hz
2-3
1751
1
1751hz
3-4
2663
0
2663hz
4-5
353
0
353hz
5-6
1017
1
1017hz
6-7
3436
1
3436hz
7-8
302
0
302hz
8-9
1513
1
1513hz
9-10
570
1
570hz
Here is the explanation of this table, the motor keeps rotating in clockwise direction during 0~1s. When the time reaches 1s, the motor just complete the rotation of 923 pulse signals. And then the motor starts to rotate in counterclockwise direction during 1s~2s. When the time reaches 2s, the motor just complete the rotation of 3540 pulse signals. So we can see that the rotating speed of motor in 0~1s is different from the speed in 1s~2s. Namely, the frequency of pulse signal in 0~1s is different from the frequency in 1s~2s.
I have already use the counter output of DAQmx, it just can generate pulse signal with certain numbers and certain frequency in one time. The attachment is the vi which can generate a finite digital pulse train from a counter output channel and the frequency, duty cycle, Initial delay, and Idle state are all configurable.
How can I continuously generate a series of pulse train with varying numbers and frequencies during a certain period of time.
Thank you very much for your help!
Solved!
Go to Solution.
Attachments:
Counter - Finite Output.vi ‏57 KB
Finite output.png ‏13 KB

Hi aacuna,
Thank you for your reply!
I already checked that article. The frequency in that Vi can change dynamically, but the method of adjusting the frequency is ‘manual’ .Below is the front panel of that Vi. When the Vi is running we can change the frequency by press the button.
Time(s)
Frequency
0-1
923hz
1-2
3540hz
2-3
1751hz
3-4
2663hz
4-5
353hz
5-6
1017hz
6-7
3436hz
7-8
302hz
8-9
1513hz
9-10
570hz
Thank you！

Sound output with sound and vibration toolkit

Hi!
Wants to setup a system with two microphones that are connected to DAQ NI USB-9162 and perform a Frequency Response between them where one of them has the undisturbed signal (close in space) and the other the respons (further away). My problem is that I want to send out noise or multitone through labview to an amplifier with internal D/A connected to my USB (turns up as a soundcard in windows). I can not find this option to send out sound in the sound and vibration tool.
Using evaluation version of both Labview and the sound and vibration kit since I wanted to see if it was possible to perform this application with Labview. Feels close but lack the success still.
Best regards
Fredrik

Fredrik76 wrote:
Ok so I have to use the Labview interface instead of the sound and vibration toolkits interface? I see the sound out option in Labview where my card apperas but not in the sound and vibration toolkits interface. Guess I have to add the options for the analysis manually then?
It should be easy to do some modifications if you have some code doing what you want, but not with a sound card. If you have some code just post it. So I can take a look. Note that sound cards do not support all sample rates. But all sound card do support 44100 Hz as sample rate. And newer sound card also higher rates. And how good your setup will be. Will depend somehow on your sound card. But by all means give it a try
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Using iMac sound output with JBP 'Creature' speakers

Am trying to install a pair (set actually) of JBL Creature speakers. I am using the iMac digital audio output which I understand should also serve as an analogue output. However, the sound out of the JBLs is quite muffled, tinny, and low, which leads me to believe that something is wrong. Any clues?
Not sure how Mac Audio outputs work.

HI,
I have the Creature with satellites. One of your satellites should have a plus and a minus sign. That is volume. The bass and treble are on the Creature subwoofer.
The outlet you use on your Intel based iMac is the one closest to the right side on the back.
Make sure your cables are seated properly. They connect "just right". I have found the Creature to have superior sound for such a small system.
More help here. http://www.methodshop.com/gadgets/reviews/jblcreature2/index.shtml
Carolyn

No choice of sound output with lightning HDMI adapter connected.

When the lightning HDMI adapter is plugged in, sound is piped through HDMI but I want the option to have the sound come through the iPad speaker, because the mini projector I use has only a 1 Watt speaker built-in, and the iPad sound is louder. Anyone figure out how to choose speaker output?

I don't believe that is possible.
Does the projector have an audio output jack to connect powered speakers?

Better estimation of phase difference between two signals with variable frequency!

Hello LabView Gurus,
Being a power engineer and having just a little knowledge of signal processing and labview, I have been pulling my hair out for the last couple of days to get a better estimation of phase difference between two signals.
We have two analog voltage signals; 1. sine wave (50Hz ± 1Hz) and 2. a square wave with exactly half of sine wave frequency at any time.
At the starting point of operation (and simulation/acquisition) both signals will have no phase difference. However, the square wave's frequency changes unpredictably for a just a few millisecond but then it gets synchronized with sine wave's frequency again. This means that the square wave will be phased out from its original position. The task of the labview is to find the phase difference between the two signals continuously.
My approach to determine the phase difference is to measure the time when sine wave crosses zero amplitude and the time when the very next square wave changes amplitude from zero volts to +ve voltage (I have a 0.5volts threshold just to avoid any dramas from small line noise). The difference between these times is then divided by the time period and multiplied by 360 to get this phase difference in angles.
As this part is just a small block of a big project, I can only allow 5000Hz sampling rate each for both signals. I read 500 samples (which means I read data from 5 cycles of sine wave and 2.5 cycles of square wave).
Now the problem is, as long as the frequency of sine wave stays constant at exactly 50Hz, I get a good estimation of the phase difference but when the frequency changes even a little (and it will happen in the real scenario i.e 50Hz ± 1Hz and the square wave's frequency is dependent of sine wave's frequency), the estimation error increases.
I have attached my labview program. From front panel, you can set the phase of square wave (between -180 and 0) and you should see the labview's calculated phase in the indicator box named 'Phase'. Then you can press 'Real Frequency' switch that would cause the frequency to change like it would in real operation.
You can observe that the estimation error increases after you push the button.
All I need to do is to reduce this estimation error and make it as close to the actual phase difference as possible. Any help would be greatly appreciated.
I am using LabView 2009 for this task.
The application is for electric machines and the stability/performance of machines under different faults.
Thank you for reading this far!
Regards,
Awais
Attachments:
v603.png ‏320 KB
v603.vi ‏186 KB

Jeff Bohrer wrote:
Basic math gives me a bit of pause on this approach. You are sampling at 50 times the frequency of interest so you get 50 samples per cycle. your phase resolution is 1/50th cycle or 7.2 degrees +/- noise. You will need to samlpe faster to reduce phase resolution or average multiple readings (at a time cost that is signifigant)
Jeff- (Hardly Working)
I am sampling at 100 times the sine wave's frequency and 200 times the square wave's frequency. Increasing the sampling rate completely solves my problem. But since I am acquiring several other inputs, I cannot afford a sampling rate higher than 5kHz.
F. Schubert wrote:
I'm not a signal processing expert, but here my basic understanding.
If you simulate sampling with 5kHz and a frequency of 50 Hz (and both are 'sync' by design), you always get an exact 5 periods. Any variation of your signals frequency gives you a propability to get 4 or 6 'trigger' events. That's an up or down of 20%!
The one measure to reduce such problems is using 'window functions'. They don't fit your current approach (counting instead of a DSP algorithm), so this needs to be reworked as well.
My approach would be to use the concept of a Locki-In amplifier. You need to phaseshift your ref-signal by 90°. Then multiply your measurement signal with the ref signal and the phase shifted ref signal. The obtained values for x/y coordinates of a complex number. Calculate the theta of the complex number (with the LV prim). Feed this in a low pass filter.
The trick on this is, that the square wave has harmonics in it, in this you are interested in the second harmonic which is the sine wave.
To get rid of the effect that the sync between sampling rate and ref signal frequency gives an error, you then can use the window I mentioned above (place it before the lock-in).
For a design that really plays well, use a producer-consumer design pattern to get the calculations done in parallel with the DAQ.
I suggest you to check on wikipedia for some of the keywords I mentioned. Go also for the external links which lead to great tutorials and AppNotes on the signal processing basics.
Sorry, it's not a simple solution I offer and we will have quite some conversation on this forum if you follow this path. Maybe someone else knows a simpler way.
Felix
www.aescusoft.de
My latest community nugget on producer/consumer design
My current blog: A journey through uml
An interesting view. the sine wave can indeed be looked as a second harmonic of the square wave. I will implement your idea and get back to you as soon as I get some results. But since I have very limited knowledge of signal processing, it might take me a while to get my hear around the solution you mentioned.

Usb sound output with integrated 100W amplif

Hi,
I would like to see a extensible USB audio device with integrated 00W amplifier. I should be able to connect for every room in my house such a device. The audi source schould not be fixed to one computer. I should be able to select different audio sources in a network environment.
or
does something like that already exist ?

I saw a device the other day, and I can't remember the name unfortunately, that allows you place multiple wi-fi boxes in various rooms through the house. I believe the box also had an amp etc. You can then choose to play different music in the different rooms. Hopefully someone else can recognise it

Need to generate a continuous sine signal to sound card output with volume control

I need to create a vi to generate a sine wave signal to sound card with Labview 8.5.
there is an example written with lvsound2.llb for it. but the volume control sucks. it has only 0~100 integer control and is not accurate enough for my application.
some one mentioned Contineous Sound Output.vi in Example search. But I could not find it any more in Labview 8.5.
Anyone could help? Appreciate it.

Dumbhead wrote:
there is an example written with lvsound2.llb for it. but the volume control sucks. it has only 0~100 integer control and is not accurate enough for my application.
Could you please explain in greater detail what you mean by "the volume control sucks"? The volume control is a Double Precision Float data type not an Integer. By accuracy, do you mean that you are trying to get greater precision? If you mean that the example simply is not loud enough at the maximum volume, you can increase the amplitude of the sine wave. I hope that helps!
Mike Lyons
National Instruments
http://www.ni.com/devzone

Is there any way to spool with variable column size?

Hi, I'm spooling to a CSV file with the following script (the real SELECT is different but similar, Oracle 10.2.0.3.0):
SET COLSEP ';'
SET FEEDBACK OFF
SET LINESIZE 2000
SET PAGESIZE 0
SET TERMOUT OFF
SET TRIMSPOOL ON
SET VERIFY OFF
SPOOL test.csv REPLACE
SELECT 'COLUMN1', 'COLUMN2', 'COLUMN3' FROM dual UNION ALL
SELECT 'value1', NULL, NULL FROM dual UNION ALL
SELECT 'value2', NULL, NULL FROM dual;
SPOOL OFF
EXIT SUCCESS COMMITThis produces the following output:
COLUMN1;COLUMN2;COLUMN3
value1 ;       ;
value2 ;       ;Is there any way to get the following output with variable column size
COLUMN1;COLUMN2;COLUMN3
value1;;
value2;;I've tried SET NULL '' but I see no difference. Thanks in advance!
Markus

In short, No, because SQL*Plus is laying out the data in columns.
You could either combine the data into a single column by concatenating as strings or use some other method e.g.
As sys user:
CREATE OR REPLACE DIRECTORY TEST_DIR AS '\tmp\myfiles'
GRANT READ, WRITE ON DIRECTORY TEST_DIR TO myuser
/As myuser:
CREATE OR REPLACE PROCEDURE run_query(p_sql IN VARCHAR2
                                     ,p_dir IN VARCHAR2
                                     ,p_header_file IN VARCHAR2
                                     ,p_data_file IN VARCHAR2 := NULL) IS
v_finaltxt VARCHAR2(4000);
v_v_val     VARCHAR2(4000);
v_n_val     NUMBER;
v_d_val     DATE;
v_ret       NUMBER;
c           NUMBER;
d           NUMBER;
col_cnt     INTEGER;
f           BOOLEAN;
rec_tab     DBMS_SQL.DESC_TAB;
col_num     NUMBER;
v_fh        UTL_FILE.FILE_TYPE;
v_samefile BOOLEAN := (NVL(p_data_file,p_header_file) = p_header_file);
BEGIN
c := DBMS_SQL.OPEN_CURSOR;
DBMS_SQL.PARSE(c, p_sql, DBMS_SQL.NATIVE);
d := DBMS_SQL.EXECUTE(c);
DBMS_SQL.DESCRIBE_COLUMNS(c, col_cnt, rec_tab);
FOR j in 1..col_cnt
LOOP
    CASE rec_tab(j).col_type
      WHEN 1 THEN DBMS_SQL.DEFINE_COLUMN(c,j,v_v_val,2000);
      WHEN 2 THEN DBMS_SQL.DEFINE_COLUMN(c,j,v_n_val);
      WHEN 12 THEN DBMS_SQL.DEFINE_COLUMN(c,j,v_d_val);
    ELSE
      DBMS_SQL.DEFINE_COLUMN(c,j,v_v_val,2000);
    END CASE;
END LOOP;
-- This part outputs the HEADER
v_fh := UTL_FILE.FOPEN(upper(p_dir),p_header_file,'w',32767);
FOR j in 1..col_cnt
LOOP
    v_finaltxt := ltrim(v_finaltxt||','||lower(rec_tab(j).col_name),',');
END LOOP;
-- DBMS_OUTPUT.PUT_LINE(v_finaltxt);
UTL_FILE.PUT_LINE(v_fh, v_finaltxt);
IF NOT v_samefile THEN
    UTL_FILE.FCLOSE(v_fh);
END IF;
-- This part outputs the DATA
IF NOT v_samefile THEN
    v_fh := UTL_FILE.FOPEN(upper(p_dir),p_data_file,'w',32767);
END IF;
LOOP
    v_ret := DBMS_SQL.FETCH_ROWS(c);
    EXIT WHEN v_ret = 0;
    v_finaltxt := NULL;
    FOR j in 1..col_cnt
    LOOP
      CASE rec_tab(j).col_type
        WHEN 1 THEN DBMS_SQL.COLUMN_VALUE(c,j,v_v_val);
                    v_finaltxt := ltrim(v_finaltxt||',"'||v_v_val||'"',',');
        WHEN 2 THEN DBMS_SQL.COLUMN_VALUE(c,j,v_n_val);
                    v_finaltxt := ltrim(v_finaltxt||','||v_n_val,',');
        WHEN 12 THEN DBMS_SQL.COLUMN_VALUE(c,j,v_d_val);
                    v_finaltxt := ltrim(v_finaltxt||','||to_char(v_d_val,'DD/MM/YYYY HH24:MI:SS'),',');
      ELSE
        v_finaltxt := ltrim(v_finaltxt||',"'||v_v_val||'"',',');
      END CASE;
    END LOOP;
-- DBMS_OUTPUT.PUT_LINE(v_finaltxt);
    UTL_FILE.PUT_LINE(v_fh, v_finaltxt);
END LOOP;
UTL_FILE.FCLOSE(v_fh);
DBMS_SQL.CLOSE_CURSOR(c);
END;This allows for the header row and the data to be written to seperate files if required.
e.g.
SQL> exec run_query('select * from emp','TEST_DIR','output.txt');
PL/SQL procedure successfully completed.Output.txt file contains:
empno,ename,job,mgr,hiredate,sal,comm,deptno
7369,"SMITH","CLERK",7902,17/12/1980 00:00:00,800,,20
7499,"ALLEN","SALESMAN",7698,20/02/1981 00:00:00,1600,300,30
7521,"WARD","SALESMAN",7698,22/02/1981 00:00:00,1250,500,30
7566,"JONES","MANAGER",7839,02/04/1981 00:00:00,2975,,20
7654,"MARTIN","SALESMAN",7698,28/09/1981 00:00:00,1250,1400,30
7698,"BLAKE","MANAGER",7839,01/05/1981 00:00:00,2850,,30
7782,"CLARK","MANAGER",7839,09/06/1981 00:00:00,2450,,10
7788,"SCOTT","ANALYST",7566,19/04/1987 00:00:00,3000,,20
7839,"KING","PRESIDENT",,17/11/1981 00:00:00,5000,,10
7844,"TURNER","SALESMAN",7698,08/09/1981 00:00:00,1500,0,30
7876,"ADAMS","CLERK",7788,23/05/1987 00:00:00,1100,,20
7900,"JAMES","CLERK",7698,03/12/1981 00:00:00,950,,30
7902,"FORD","ANALYST",7566,03/12/1981 00:00:00,3000,,20
7934,"MILLER","CLERK",7782,23/01/1982 00:00:00,1300,,10The procedure allows for the header and data to go to seperate files if required. Just specifying the "header" filename will put the header and data in the one file.
Adapt to output different datatypes and styles are required.

Variable frequency driver control by analog voltage output

I need to drive a variable frequency inverter to drive a compressor and I am too newbie in this kind of works. The inverter can be controlled by analog voltage input and I have NI9188 and NI9263. I have never used analog output modules before and I have some doubts about them.
1. how could I drive the motor continuously?
2. do I have to use device internal clock or software clock?
3. in some examples, a waveform is generated and sent to the write daqmx vi to generate analog output. do I need to do it? (By the way, I prepared a control subVI to send the required input voltage data to the write daqmx.)
4. It is talked about pulse with modulation sth like that. What is it exactly? Do I need to concern about it?
I need to drive the compressor with two temperature feedback so I prepared a huge program by master/slave design pattern. I attached it. Does this program work? I do not want to harm the motor so I could not test it.
Egemen
Attachments:
Control Project.v0.zip ‏2672 KB

Hi newbieeng,
I hope you are doing well today! I can help answer questions, but I can't debug a program and say whether it will work or not.
A lot of these question depend on your specific application and can't be answered in general. I would take a look at the specifications document for the NI 9263 and also the specifications for your motor. I'm including a link below to the specifications document for the NI 9263. I would highly recommend looking at the current draw of the motor. The most the NI 9263 can drive is 1 mA per channel.
OPERATING INSTRUCTIONS AND SPECIFICATIONS
NI 9263
http://www.ni.com/pdf/manuals/373781e.pdf
As far as hardware timing vs. software timing, it really depends on what you want to do. Hardware timing provides most more precise control over the timing of the output.
From one of the subVIs you provided, there seems to be a mention of a relationship between frequency and RPM. Is this representative of a control signal the motor expects? Is this the pulse width modulation (PWM) you were talking about? If your motor requires PWM, then you will need to send it the appropriate signals. I'm attaching a link below to an example regarding PWM.
Pulse Width Modulation
https://decibel.ni.com/content/docs/DOC-17537
Hopefully some of this is helpful!
Josh B
Applications Engineer
National Instruments

Help with sound output via HDMI

My sound output via HDMI suddenly stopped worked to my TV set. I've reset the PRAM and the SMC. When I go to sound control panel for sound output, the setting is on Headphones. When I try to change it to SV420XVT1A (the tv) it immediately jumps back to headphone. I also tried a different HDMI cable to a different HDMI port on the tv. This just started after the Mac Mini had been turned off for a week. I'm using Lion and it's an 8GB Intel Core 2 Duo. Any suggestions?

I actually discovered this answer about 5 minutes after I posted the question. Thanks a lot for your help!! Funny thing is that it had worked all along with something plugged in the headphone jack but suddenly stopped working. Again, thanks!

How to fire sounds with various frequencies

Hello,
I'm willing to create a program to check how people hear sounds and plot curves based on audible frequencies.
Are there java classes to play a sound at a particular frequency. ?
I'm running Windows XP on a PC with just speakers and earphone.
Thanks to point me to the good package (if it exists) or to provide me with a sample code.

There's a bug in the forum software that's inserting
a space when displaying long urls, like here:
c2coff=1& q=test
Darn, didn't notice the blanks. Usually they're more obvious.
b) searching for the keywords you used doesn'tbring
up any code as far as I saw. Funnily, though, it
returns this thread...Then I guess that you skipped over this one:
http://www.softsynth.com/jsyn/tutorial/
(#10 in the list)
and this one:
http://javaboutique.internet.com/audio/
Very probably. All I saw on my list were online hearing tests, but no code. :) Thanks.

No sound output in games with "advanced effects"...Audigy 2 ZS...HEL

<SPAN>Hi there.... I have a soundblaster audigy 2 ZS installed with the drivers and EAX console. I have noticed that if I enable "advanced EQ" in the "enable audio effects" menu, i get no sound output from games or other software, although windows noises and sound files play fine!!If i use other features from the "enable audio effects" menu such as "environment" then the sound works fine....Surely this is not meant to happen?Anyone else check this for me?

*bump*

Continous sound output with variable frequency?

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