Very Low Frequency Converter

Hello
There are numerous devices that transmit audio via very low frequency (VLF). I quote:
"Carrier current devices are a combination of technologies. They are a cross between
wired microphones and subcarrier transmitters. The only difference is that the signal
is not transmitted via radio waves, but rather through a wire pair. A person cannot
accidentally intercept or detect a carrier current signal by simply tapping into a wire
like with wired microphones. A carrier current device works by picking up room audio
through a microphone. The signal from the microphone is then modulated by a low
frequency circuit, which produces a carrier current signal at approximately 100–200
kHz. A common example of carrier current devices are the newer wireless telephones,
intercoms, or baby monitor type devices that plug into the electric socket and use the
pre-existing wiring rather than having wiring run all over the house. A special circuit,
which can demodulate the low frequency signal, is used as the receiver.
Only a sophisticated receiver with a low frequency probe can detect this sort of device."
Can Audition modulate the signal back into an audible range?

At 100kHz, yes it's true - you'd need an audio interface running faster than twice the modulating frequency to demodulate all of the sidebands correctly. Not it's not absolutely impossible to do this, but it's a very expensive way of doing what hardware does very cheaply! To do this in Audition, you'd need initially to get the modulated carrier in (so if it was 100kHz, that would require an interface that runs at 200kHz to satisfy the Nyquist requirement) and all of the currently available crop appear to stop at 192k), and then run the signal through an envelope detector to capture the sideband signals. Problem is that Audition doesn't have the envelope detector, so you're stuffed anyway, even if you used 100Hz. 100Hz would give you all sorts of problems on top of this, because you'd have to use FM and then you'd have to deal with all of the reflected sidebands.
So to state this baldly, Audition is an audio editing program, not a radio receiver!
If you want to read more about this (and I should warn you that this gets complicated) then here are a few links:
Demodulation - Wikipedia, the free encyclopedia
audio-rate frequency modulation
http://www.secretmango.com/jimb/Whitepapers/radio/radio.html

Similar Messages

  • "How to measure very low frequency?​"

    I'm using Labview 5.1 and a data acquisition card PC-LPM-16PNP. How can I measure a low frequency signal like the human pulse of 72 pulses/min or 1.2 Hz. Thanks.

    Virgo,
    When measuring very low frequencies, I recommend measuring the period between pulses and converting to frequency.
    To measure the period, there are a few methods. For something like a heartbeat, I would probably use a peak detector to locate the peaks and measure the distance between the peaks. For other signals, it may be easier to locate zero crossings. Sometimes filtering or smoothing is required to remove noise and improve the results.
    Bruce
    Bruce Ammons
    Ammons Engineering

  • Why am I getting very high values for the very low frequency region of a random signal?

    I am tyring to produce a power spectrum graph for a Tachogram data, related to Heart Rate Variability Analysis. This data can be thought to be as a random signal, but has a frequency spectrum range of 0 - 1HZ.
    The problem that I am facing is, I am getting very high values for very low frequency region, closer to DC value. Even the DC value is really high, in the range 10^8. It is suppose to be a low number. Any suggestions would be appreciated.
    Thanks.

    Here is what my work is all about. I am trying to develop a software for Heart Rate Variability analysis. I am not sure if you are aware of heart beat waveforms, they are bunch spikes, occuring at irregular intervals. We have to do analysis on this waveform. How ? First we have to create a plot called Tachogram. This is done by, for example, let's name the first spike R1, the second one R2, and the third one R3, and so on. This is how the co-ordinate points are created. (R1, R2-R1), (R2, R3-R2), and so on...
    Here R1 is the time instant at which the spike R1 occurs, and R2-R1 is the time difference between these peak occurences. So, if we plot these values, then it gives us one like what you saw on your time domain plot. What we have to do with this signal is, we have to apply FFT technique to produce its power spectrum for analysis. Generally the frequency range of the spectrum goes from 0 - 0.5 Hz. They part this range into VLF, LF, and HF, and analyze how much power is distributed in these regions. The distribution would tell you if you were stressed out,or have worked out, things like that. Normally, you can view spikes around LF, HF region. If you had viewed the power spectrum for the data that I sent, then that's what it pretty much looks like. I think I have said a lot about it now. Hope you get it.
    We were talking about the "other component", I can't really see where it lies, must be in the 60-80 sec. range in the graph. If you look at the spectrum, I am getting a huge value around 0.0033 Hz, how do I minimize or remove this ?
    Also, I am using Unevenly Spaced Signal Spectrum.vi, I am not sure what excactly is the unit for the Power Spectrum output. Technical Papers denote, Periodogram algorithm used by this VI, gives Power Spectral Density ? If that's right, then the unit is V^2/HZ.
    Please advise.
    Again, I should thank you for your time and effort.
    I have to meet a deadline this week in finishing this project, and your help is immense to me at this point.
    Thanks & Best Regards,

  • Very low frequency caused by sample frequency in FFT analog input?

    I'm measuring a very low frequency on my analog input, this frequency is in connection with the sample frequency of the Analog Input. At a sample frequency of 1000Hz I see a frequency of 0,05Hz in my FFT, at a sample frequency of 500Hz I see a frequency of 0,02Hz.
    Attached is a screenshot of an example how I see this very low frequency.
    My hardware: NI USB 6008 --> measuring on AI-0 (in this example the input is unwired). But in my real measurement I see the same FFT + signals I want to see (about 2 Hz).
    In my real measurement I windowed the FFT (1-3Hz) so I see only the FFT I want to see. But I suspect that my complete signal moves along with this very low frequency of 0,05Hz. I saw this in my measerement.
    What did I do wrong?
    Attachments:
    screenshot.JPG ‏66 KB

    First, do you live in Europe? If so, that 50Hz could be power-line pick up.
    Antialias filtering must be done in hardware before the DAQ. Because of the way aliasing works if you have sampled the signal it's already too late, you're hosed and no amount of digital filtering can remove the aliased signal. In terms of filter specifications, the filter cutoff needs to be at twice the highest frequency you are interested in seeing. For example, if you are looking for signals in the 2- to 5-Hz range, your antialiasing filter should cutoff at around 10Hz.
    Obviously good signal management is also needed: shielding, appropriate signal termination, proper lead dress and spacing from known noise sources, etc...
    Mike...
    PS: There were no attachments to your last post.
    Certified Professional Instructor
    Certified LabVIEW Architect
    LabVIEW Champion
    "... after all, He's not a tame lion..."
    Be thinking ahead and mark your dance card for NI Week 2015 now: TS 6139 - Object Oriented First Steps

  • 27" iMac screen low frequency humming + moving too freely on the base

    Got a new 27 incher on tuesday after fedex lost the first one That's a different story.
    The problems
    1. The huge iMac moves a tad bit too freely on its base and when i try to tilt it to an angle, it doesn't stay at that angle it trys to go tilt back down ... maybe its gravity.. i don't know...
    2. Very very low frequency humming noise from the iMac top left corner... its out when it goes to sleep. not the normal HDD/fan noise. I have a fan app, its not the fan... but its really annoying when working on it ...
    i've seen a couple of posts about this. i'm taking this 27 incher to the store tomorrow... lets see how the geniuses deal with it. i hope they acknowledge this problem. does anyone else with the 27" have this problem?

    Took it to the apple store, some genuises looked at it, confirmed it and sent me another one

  • Low frequency measuremen​t from Parallel Port

    Hi there...
    I need to calculate the "on" and "off" time and duty cycle in pulse form from a parallel port. By making the circuit in 5 or 0 V, I just simply put it in my parallel port. The problem comes when I need to measure a very very low frequency. In this case, I want to measure the duty cycle from my operated refrigerator. I need to know when the thermostat goes "on" and when it comes to "off". In my experience, the thermostat will be "on" in about 5-10 minutes and "off" in about 20-30 minutes. So, the pulse might be take for a long periode each.
    I've tried with Timing and Transition Measurement wizard or even by using Pulse Measurement.vi which is included in Waveform Measurement category. It only works for 2 Hz and . If I try to set it with 1 Hz or below, it comes the message :
    "Error -20308 occurred at Timing and Transition Measurements -> Untitled 1
    :4"  (waveform index 0 of 1)
    Possible reason(s):
    Analysis:  The waveform did not cross the mid reference level enough times to perform this measurement. Check the signal length, reference levels, and ref level units."
    Could someone help me please ?
    Regards,
    Ricki

    here is a quick shot to give you an idea
    Greetings from Germany
    Henrik
    LV since v3.1
    “ground” is a convenient fantasy
    '˙˙˙˙uıɐƃɐ lɐıp puɐ °06 ǝuoɥd ɹnoʎ uɹnʇ ǝsɐǝld 'ʎɹɐuıƃɐɯı sı pǝlɐıp ǝʌɐɥ noʎ ɹǝqɯnu ǝɥʇ'
    Attachments:
    port logger.vi ‏22 KB

  • Measurment through a vibration sensor interfaced with Ni-PXI is very low & less responsive. I wana know any frequency limitations is there for PXi or the system uses any clipping ckts inside ???

    Measurment through a vibration sensor interfaced with Ni-PXI is very low & less responsive. I wana know any frequency limitations is there for PXi or the system uses any clipping ckts inside ???

    It would be helpfull to know what PXI hardware and what sensor you are using.
    Up to now I only found mechanical limitations ....
    (Creating a 100kHz sinus exitation with 160nm amplitude need more than 63km/s² (>6400g) acceleration  )
    Greetings from Germany
    Henrik
    LV since v3.1
    “ground” is a convenient fantasy
    '˙˙˙˙uıɐƃɐ lɐıp puɐ °06 ǝuoɥd ɹnoʎ uɹnʇ ǝsɐǝld 'ʎɹɐuıƃɐɯı sı pǝlɐıp ǝʌɐɥ noʎ ɹǝqɯnu ǝɥʇ'

  • Dirty mid-range frequency and very low vocal

    I noticed this problem after I installed and began using Bose USB speakers on my Windows XP. In my iTunes, many of the songs have staticy mid-range frequency sound and vocals are very low and difficult to hear. These songs include songs that were ripped directly from marketed CDs or downloaded from iTunes Store. My equalizer is set to Flat and messing with the equalizer didn’t help as it made the songs even more staticy. I don’t have this problem on QuickTime, Windows Media Player, Realtime, nor on iPod. Only on iTunes. I want to use iTunes since I find it more convenient and easier to use than any other players that I know.
    Can this be fixed? Please help.
    iTunes 7.1.1.5 Windows XP

    btw, I'm using my built-in laptop speakers (Altec Lansing); so it's not a Bose thing.

  • Low frequency measurement (1 to 5 Hz) issues - Reg

    Hello All,
    i am facing a problem in measuring Low-Frequencies in the range of 1-5 Hz. I am using VIs used for pulse-width measurement and the card is PCI-6031E. i am generating frequency using PCI-6013 card.
    Actually i am trying to develop an application for measuring the speed (RPM) of the motor which runs very slowly. so if  i miss one pulse  per second also, when i multiply that with 60 it will be big error.
    So if any can give me a solution that will be big help to my issues.
    Thanks.

    Hello sathiskumar,
    I'm not sure what you mean by missing a pulse but I think what is
    happening is that your counter is rolling over and starting to count
    from zero again.  Every time the counter rolls over, it will
    output a pulse.  You can connect this signal to another counter
    and count how many times your first counter has rolled over.  That
    way, if you don't receive a pulse on your first counter before your
    counter rolls over, the second counter will have a record of how many
    times this has happened.
    I hope this helps!
    Eric
    DE For Life!

  • I do a radio show, when I pre-record i get a very low end hiss, almost sounds like gain is too high, but its not. If Im live to the station its fine. It only happens when I record. Is there a way to set audition NOT to record that low end sound?

    I use an Allen & Heath broadcast board. When I record to my computer for later broadcast there is a low end hiss/hum barely noticeable when im speaking BUT very prominent when there is a quiet moment.
    Sounds like gain may be tooo high, BUT its not.  When I do a LIVE broadcast using same equipment all is fine!
    Thoughts

    Well, several thoughts here...
    First, what are you using to get from the A&H mixer into your computer and is it in the chain when you broadcast live?  I ask because a lot of built in sound cards are pretty noisy and that could be the source of the problem, fixable with even a very cheap external USB interface
    Second, if the hiss actually starts at the mic/mixer, I know the A&H XB series has a high pass filter on every mic channel.  Have you engaged that?
    That said, it's pretty rare for hiss to be such a low frequency that the high pass filter will kill it.  You may get on better with the Noise Reduction feature in Audition, applied after the recording.
    Anyhow, to repeat how I started, what's your signal chain when recording and how does it differ from when you go live?

  • What is the max. cable length that the Differential TTL signals can run in both at high and low frequencies?

    Hello,
                I want to now the max. cable length that the Differential TTL signals can run in both at high and low frequencies.

    That is very dependant on the type of cable, its construction, and inherent impedance and capacitance. This may be of use:
    http://beiied.com/PDFs2/SSI_14-15-bit-encoder_addendum.pdf
    -AK2DM
    ~~~~~~~~~~~~~~~~~~~~~~~~~~
    "It’s the questions that drive us.”
    ~~~~~~~~~~~~~~~~~~~~~~~~~~

  • Rendered video very low quality

    My camera shoots in MTS format. I converted the file to mp4 and edited it. I then imported that footage into After Effects and when I tried to render the video it was very low quality. I made sure the composition settings and render settings matched that of my video. I tried saving it as H.264, mov, avi, m4v, and mpeg2 non of which were good enough quality to use. The avi format came close in quality for the video but everything seemed to move in slow motion. The audio played perfectly though. I also tried rendering video directly from the MTS file and it still didn't work.
    Is there a specific way I should convert or encode the file? I tried some different settings within Adobe Media Encoder but when I brought the file into AE and tried to render I got the same issues of it being low quality.
    This is for a client so it's very important that I get the needed effects added to this video. Thanks in advance for any help!

    I understand how After Effects rendering works and different file types. I just wanted to try rendering in different formats just make sure I hadn't missed something. I figured something out that came close to the quality I need so I guess it will have to work.
    My original MTS file is 1920 x 1080 and 29 fps.
    I rendered it with these settings originally
    NTSC, 1920 x 1080
    29.9 fps
    Bitrate CBR 30 Mbps
    The file came out choppy and noticeably lower quality.
    The last time I rendered it with these settings:
    NTSC, 1280 x 720,
    59.94 fps
    Bitrate CBR 20 MBps
    The quality was much much better. It still had a little distortion with a lot of movement but it was much more bearable so I guess that's what I'm going with. If anyone knows of any better settings it would be greatly appreciated! Thanks.

  • ADAT into Logic Express, VERY low signals recorded

    I tried Garageband, same problem, so now I am trying Logic Express.
    When transferring tracks from an ADAT, through Alesis io/14, into Logic, the resulting sound files captured are VERY low signal. This should be a digitally ACCURATE TRANSFER. The Alesis montoring software on the mac senses the correct signal, the signal meters show correct levels (and sounds great), but Logic hears and records very low signals.
    I note the Sound midi pref window only shows 6 channel 32 bit. It won't allow me to change to 8 channel 16 bit. Possible problem or am I chasing ghosts? I exhausted all my ideas.
    Any thoughts?
    T

    Ropher wrote:
    Discovering I was unable to import a ".au" file into Logic Express (or Garage Band) I read it into Sound Studio and exported it in a variety of formats. I can now successfully drag to Garage Band files in formats .aif .aifc .sd2 and .wav and get the expected result.
    Unfortunately however, I only get the expected result importing these into Logic Express for the .aifc format which plays back at the expected speed with a low-pitched sound (but which doesn't display an uncompressed waveform). Importing the .aif .sd2 or .wav format files however results in the imported file being squeezed into a very short duration with a correspondingly undesirable very high pitch.
    Is there some setting or magic incantation I need for reading in audio files to Logic Express without it getting squeezed?
    Thanks!
    --Roph
    Yes. you need to ascend to level 5 before we give that secret incantation.
    Really, you need to use a third party converter software, if you are going to use .au files.
    Logic is happiest with .aiff, .wav files, but not much else.
    Cheers

  • Audio outputting very low

    I'm outputting to flv's and there is sound but very low. I converted three files the same exact way and only one came out the way it should have. The only difference I see is I used the real camera audio on the one that sounds like it should and wav files for the other two that came out low. The wav files are clean and at the same level as the original.
    The thing that baffles me is that I exported all 3 to QT reference files and they all sound great.
    I have converted to another format and still low. I have also created a new QT reference and still the same problem.
    Anyone know why and how to fix this...why does it look and sound great in my QT reference but not outputting that way?

    I'm just going to make a wild guess here.
    Sometimes, if you have two audio tracks that are out of phase,
    when you combine them, the result is very quiet or disappears.
    If, for example, if the left and right channels are out of phase,
    you will hear them OK in stereo, but if they are both panned center,
    and there is a phase problem, you could be losing a lot of level.
    If you just use one track and mute the other, is the problem still there?
    Does the problem only happen when you're using both tracks?
    That would be a phase problem.
    In that case, you could try just use one track and pan it center.

  • Very Low Bit Rate!!!

    Ok, iTunes is not the best program of the computer planet music, far away…
    All music sites offering the possibility to choice the format and downloading with a quality what we wants, no iTunes Store, WHY? I have professional equipment and disgust me to listen my music in 256kb very (LOW BIT RATE).
    I spend the max in the MacBook Air (the best configuration), the must of sound card etc. for what??? It’s time to change and make a true revolution on iTunes and iTunes Store. For ex: when we listening one song on iTunes, the folder don’t come to the principal window if you are in artist’s mode! We need to search in the library to find the folder and get information or modify something! 

    Is the file stored in iTunes as a music file?
    If so then the one way of getting it on your phone would be to convert it to a podcast or audiobook, which I believe will work.
    It isn't a full conversion as such, all you do is change a tag in "get info".
    Another way would be to choose "create AAC version" which should make it eligible.
    I understand why you don't want to do this sort of messing around, but nfortunately at this stage in the release there are still lots of these sort of issues, most of which will likely be resolved in the next few updates.
    If you don't want to do those steps then another option would be to turn match off until these issues get fixed. Obviously, we have no idea of timescales (or if that particular issue will ever get addressed). There are enough people being affected by it that I'd be surprised if it isn't a reasonably high priority though.

Maybe you are looking for