Record sound with mic, phase shift 180 degrees, then play?

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  I bought an ADICON SSB audio analog digital converter (ssb-SEO).

  This device supports 44.1, 48, 88.2, 96, 176,4 and 192 kHz 24 Bit Audio sample class 2.0.

  Sampling by hearing works perfectly between 44.1 and 96 kHz. Soon, I have try 176,4 or higher, hearing said that the input and output device is not the same sampling frequency.

  Saying this, the Mac output device is the build in audio and is limited to 96 kHz. In my case, I do need nit an output device, because I want only to save my music in the highest available sample rates, transfer to the PONO of Neil Young player and want to be happy.

  Can someone help me solve this problem?

  Thanks a lot for your answer. Products like AUDACITY can enjoy Audio with 192,000 Hz and they also support listen-true. I'm a fan of ADOBE and do not want to throw my up-front investments.

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• Cannot record sounds with microphone

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  I'm not familiar with this machine and I do not read German, but in the control panel Windows 7, you probably manage Audio devices, and also the Realtek HD Audio Manager.

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  Perform a procedure of similat to the Read tab and in the sounds tab choose No sounds like the sound scheme.

  Now out of this Audition CS5.5 and open in Audio hardware installation, make sure you that have selected high Ms.

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• phase shifted PWM with Ni 9401 and Crio

  Hello

  Do you have an idea of pwm shifted 180 degrees?

  (duty cycle frequency and variable difficulty)

  I tried a design, but it seems in the graphics design works on the realtime.vi but it does not work with the fpga.

  Graphic output pwm FPGA are distinguished by the real time as you can see in the pictures.

  On the other hand, VI Fpga produce two pwm, as seen in the oscilloscope when the fpga VI runs.

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  Hi Maurice

  Thank you for your help.

  Yes I want to that they will be moved with the right variable and 10 kHz. I put 49% maximum duty.

  I put the output into the same output block.

  Square wave generator does not accept 'loop' while.

  I have attached a simple FPGA project file. Could you please tell me what is my fault?

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• How to record audio with my laptop closed?

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• Transformed Hilbert phase shift of 90 degrees

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• The volume of the recorded sound is very low when played. I tried to solve the problems through Control Panel of configuration-sound options and increased the volume to 100 form my mic too. All the other volume options, it's 100 too. It has no effect. Hel

  The volume of the recorded sound is very low when played. I tried to solve the problems through Control Panel of configuration-sound options and increased the volume to 100 form my mic too. All the other volume options, it's 100 too. It has no effect. Help needed ASAP

  Hello

  Control Panel - SOUNDS - Audio peripheral devices configuration-manager

  Make sure you write down the description and model double click top - tab drivers - write version. Now
  Click on update drivers who cannot do anything like MS is far behind the certification of drivers. Then RIGHT
  CLICK above and UNINSTALL - REBOOT - this will rebuild the driver stack.

  If your driver shows 06 you have the default windows and part of the question.

  Go to the system manufacturer's website and get the latest Audio/Sound drivers - download - SAVE - go to the place where you
  Put them - right click on Reboot - RUN AS ADMIN -.

  Check the Version on the tab of the driver in the device as a sound Manager often restoring drivers, then install
  the current version. Restart and check after each try, not uncommon to have to run the pilot of 1 to 4 times or more.
  Just make sure that the version you have installed appears in the tab of the driver.

  Then navigate to the site of the manufacturer of the device and search for the drivers more recent and repeat the installation procedure.

  Look at the sites of the manufacturer for drivers - and the manufacturer of the device manually.
  http://pcsupport.about.com/od/driverssupport/HT/driverdlmfgr.htm

  How to install a device driver in Vista Device Manager
  http://www.Vistax64.com/tutorials/193584-Device-Manager-install-driver.html

  ---------------------------------------------------------

  After the updated driver.

  Click with the right button on the speaker near clock - recorders - right click in the box - check show disconnected devices
  and show disabled devices - check your micro - highlight them, then lower right properties. Do
  of course the appropriate injector is enabled and check that other settings - on the levels tab make sure MUTE is not checked
  and APPLY / OK changes.

  Also in the search for some Audio Manager Panel (perhaps named after the manufacturer of the sound driver). Mine
  is the RealTek Audio Manager, yours can be quite different name if you have one.

  How to activate Audio devices hidden in Vista
  http://www.Vistax64.com/tutorials/143447-audio-device-enable-hidden-devices.html

  Connect one microphone, music player or other audio device to your computer
  http://windowshelp.Microsoft.com/Windows/en-us/help/b0e3d7fb-1b4a-4926-8254-65da289209261033.mspx

  Problems with sound and audio or no sound - a Mr Fixit
  http://support.Microsoft.com/GP/NO_SOUND

  I hope this helps.

  Rob - bicycle - Mark Twain said it is good.

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• With the help of a Butterworth filter without phase shift

  Hello

  I found this reference to the use of a Butterworth filter without phase shift http://zone.ni.com/devzone/cda/epd/p/id/2775 but I can't open the sample with LabView 2011.  Can anyone help?  The referenced file is attached.

  Is there another or the best way to do this?

  Thank you

  Attached is a version saved in LV2009.

  In such cases, you can validate the vi to the discussion of queries VI Upconvert.

  Ben64

• Meter with two adjustable phase shift

  Hello

  In this experimental device, I have a print head a TTL pulse-controlled piezoelectric ink jet delivering uniform droplets on a surface. I use the "time" version of the counter output vi (high-/ low-time) because it allows me to very easily change the characteristics of the droplets. I use a strobe approach for imaging the droplets as they are ejected. Basically, a strobe LED light is pulsed at a frequency that exactly matches that of the inkjet printhead. A CCD camera is used in order to imager droplets, who seem "frozen" on the screen due to the stroboscopic effect. Strobe LED is triggered by a train of pulses TTL (two pulse trains come from exits of meter on my USB-6353 X Series DAQ board).

  Of course, I could trigger both the inkjet Printhead and the strobe light with the same output of counter, which would ensure that their frequencies match. But it's really nice to have a 'strobe delay' that allows adjustment of the phase shift between the strobe triggers and printhead. The hardware supplied with the print head has this feature of strobe delay as an external button. It is useful, because you can basically lead through time by turning the button and view the formation of droplets when it leaves the end of the nozzle.

  I have a vi that may trigger sometimes the printhead and the flash, but I can't understand how to adjust a phase shift between the two, while the program is running. It should be possible, but I can't get it. I would really appreciate help with this. Attached is the draft code and a diagram which may help to explain what I want to do

  Thank you very much

  -Matt

  No - forget the INITIAL DELAY.  It's only for the (first) INITIAL pulse.

  You already want to adjust the time / low-time already, no?

  So having a new control called PHASE SHIFT, from scratch.

  Have a variable called OFFSET PHASE CURRENT, from scratch.

  When the PHASE SHIFT is modified (by the user), understand the difference between where he wants to be and where you are (control - PHASE CURRENT OFFSET) and add a lot of time the low TIMES, but only during a cycle.  Basically you're stretching of a cycle.  Store the new value in the course of PHASE SHIFT variable for next time.

• Microsoft Lifecam 5000. Whenever I'm recording video with her, during playback, I hear clicking on and a beep sound when the camera focuses

  Hello, I have problems with my Microsoft Lifecam 5000. Whenever I'm recording video with her, during playback, I hear clicking on and a beep sound when the camera is focusing. If I put it to my ear, the sound is low but I can still here it when it focuses.

  Hello

  Thanks for posting in the Microsoft Community Forum.

  I understand from your description, that you get beep when the camera is focusing.

  To better understand the question, please answer the following questions:

  1. have you made changes on the computer before this problem?
  2. is it works much earlier?

  Come back with more information to help you better.

  I suggest you to download and install the latest drivers for Microsoft Lifecam 5000 from the link provided:

  http://www.Microsoft.com/hardware/en-us/d/LifeCam-VX-5000

  You can even try recording by reducing the volume on the computer to the minimum level (mute) just to check if beep occurs.

  http://Windows.Microsoft.com/en-us/Windows7/adjust-the-sound-level-on-your-computer

  For more information, see the link below:

  http://www.Microsoft.com/hardware/en-us/help/support/troubleshooting/webcam/LifeCam-Setup

  Hope the above information is helpful. If you have any question back to us, we are happy to help you.

  Thank you.

• I can't record sounds or voice with my acer M5-581TG. Why? I can listen to the music...

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  You really cannot save without a connected microphone

• How can Satellite Pro A300D-1E7 - I record sounds?

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  Thank you
  David

  Hello!

  I don't know if it would work, but you can try it and give us feedback :) m

  You have a program to record sound?
  So I can recommend the No. 23 Recorder (Google is your friend).

  Good bye

• Align the two signals and measure the Phase Shift

  Hello

  I do an experiment in which I use the NI USB-6221 DAQ card. The jury is able to make 250 k samples/second. I want to measure two voltages in a circuit and find the phase shift between them at frequencies between 1 and 10000. First I ouputted a wave sinusoidal frequency variable through the Commission and applied to a test circuit. Then I used the Board to measure the two tensions consecutively (thus reducing the maximum sampling frequency at 125 k). I used the signals align VI and measured the two phases and then calculates the phase shift (VI attached in Phase 1). It worked well for the test circuit I built in which the phase shift went way logarithmique.20 degrees ~84.5 degrees and then stabilized. At frequencies above 5 000 Hz phase shift must have remained constant, but it varies more or less 1 degree. When the phase shift is 84.5 degrees, present a degree of variability is not particularly explicit. When I asked my program on the circuit that I really wanted to measure, the phase shift went from-. 5 degrees up to about 1.2 degrees. The change in the values of phase shift at high frequencies (> 3000) was environ.2 degrees. Given the small phase shift, this variation is unacceptable. Now I tried to use a sequence to each blood individually (increase the maximum sampling frequency to 250 k) and then align the two signals and measure the phase of each shift. When I use align it and re - sample Express VI to realign the two signals, I get the message "error 20333 analysis: cannot align two waveforms with dt even if their samples are not clocked in phase." Is it possible to align two signals I describe here? I enclose the new VI as Phase 2

  Matthew,

  I think I have an idea for at least part of the problem.

  I took your program data and deleted stuff DAQ.  I have converted the Signal on the chart control and looked then what was going on with the signal analysis.

  The output of the Waveforms.vi line has two waveforms, like the entry.  However, arrays of Y in the two waveforms are empty!  It does not generate an error. After some head scratching, reading the help files and try things out, that's what I think is happening: the time t0 two input signals are 1,031 seconds apart. Since the wavefoms contains 1,000 seconds of data, there is no overlap and may not align them.

  I changed the t0 on two waveforms are the same, and it lines up.  The number of items in the tables is reduced by one. Then I increased the t0 of 0.1 seconds on the first element. The output had both greater than the entry by dt t0 t0 and the size of the arrays was 224998.  Reversing the t0 two elements shifts the phase in the opposite direction.

  What that tells me, is that you can not reliably align two waveforms which do not overlap.

  I suggest that you go to 2-channel data acquisition and that it accept the reduced sample rate.  You won't get the resolution you want, but you should be able to tell if something important happens.

  You may be able to improve the equivalent resolution by taking multiple steps with a slight phase shift. This is similar to the way that old oscilloscopes of sampling (analog) worked. Take a series of measures with the signal you are currently using.  The make enough average to minimize changes due to noise. Then pass the phase of the signal of excitement to an amount that is smaller than the resolution of phase of sampling rate and repeat the measurements.  Recall that I calculated that for a 5 kHz signal sampled at 125kHz, you get a sample every 14.4 degrees. If shift you the phase of 1 degree (to the point/mathematical simulation), you get a different set of samples for excitement.  They are always separated by 14.4 degrees.  Take another series of measures. Transfer phase another degree and repeat.  As long as your sampling clocks are stable enough so that frequency does not drift significantly (and it shouldn't with your equipment), you should be able to get near resolution of what you need.  The trade-off is that you need to perform more measurements and may need to keep track of the phase shifts between the various measures.

  Lynn

• measurement of phase shift between two periodic signals acquired

  Hello

  I don't know how to explain my problem, but I'll give it my best shot. I'm two signals from sensors in tension. Two periodic signals have the same frequency, but a different amplitude. Normally they have a difference of phase of 0 or 180 degrees. The thing I have to let labview to check is if the signals are completely in phase or out of phase (180 degrees) completely. I am acquiring the wizard scene DAQ in a while loop. Does anyone have an idea how I can do this?

  To summarize: are the two things I need to know

  -l'amplitude (maybe just pick max max min distance)

  -If both signals are moving in the same direction (when the two signals are in phase) or if they move in the opposite direction (counter phase) - the exact phase angle value is not so important

  THX

  Thanks for the reply. After a night to think about the problem, I came up with a home-made solution. I used the point by point max and min vi to calculate the distance from crest to crest of my signals. Then, I used the time derivative of point by point to calculate the variation of the signal of the two signals. If the signals are in phase (both are increasing or decreasing at the same time with the other), then the two derivatives have the same sign (accept on the summits, of which I have excluded from the comparison). If the signals are out of phase (when one increases, the other is decreases and vice versa) then the two derivatives are opposite in sign. According to this, I can get all information that I need. If anybody should know a better way to achieve the same result, please say, but to know that it seems to work.