Aud 2 NX - Cant Select 96Khz Sampling Rate/Dolby Digital Light doesn't Illuminate when playing DV

Hi,
I'm running a Audigy 2 NX (USB) on a Toshiba 240-504 Notebook through the rear USB port, the speakers I am running are Logitech X530's (A bargain if you can get them, but that's beside the point =])
I cannot select 24Bit/96Khz from the Device Control applet, I should be able to. All I can select is 24bit/48Khz.
Also, when I'm playing Dolby Digital content such as a DVD through PowerDVD, the Dolby Digital light on the card does not illuminate, is this expected behaviour? The light Illuminates if I use the product showcase demo cd though, by the way. Also, slightly off topic, does anyone know what settings I should be using for PowerDVD? Is CLMEI-2 ok?
I will post anymore information you need to know at request. Thanks very much for your time reading this.

th0r0n,
Are your notebook using USB2.0 as those options are only available if you are using USB2.0. There's a chart in the manual also that shows these. Also, the Audigy 2 NX dolby indicator will only light up if it's decoding AC3 material. If you are using PowerDVD, you will need to config PowerDVD to use SPDIF and passthrough the AC3 signal to the sound card for decoding.
Jason

Similar Messages

  • How to create a wav file from 24bit 96Khz sampling rate data

    Hi
    I am trying to make an VI which will play sound while acquiring data from PXI 4472 DAQ card.
    My sampling rate is 96Khz and PXI 4472 card is a 24bit card.
    Wave files are in 8 or 16 bit and the sampling rate is 8000, 11025, 22050 and 44100. How will I be able to play the data which I am acquiring.
    How would i normalize the data into the required format needed for most of the sound cards to play.
    Or are there any codec available in Windows XP which i call to play a 96KHz 24 bit sample
    Does anybody ever encountered this type of problem.
    Thanks in advance
    Nitin

    Whilst the 'standard' RIFF format specification usually accomodates 16 bit data, there is of course no reason that you can not create your own extension. It just won't be playable by Media player using the 'standard' installed drivers or codec. This may not be a problem....
    WAV files can and do support other formats, you just need to know how to handle them......
    There is howerver a 4GB limit (related to the pointer size in the WAV specification) which with higher bit depths on the sampling does start to become a bit of a problem.
    To give you a few samples of other types of wav files check out the following site here
    http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
    The following definitions for WAV audio formats may also be of interest here
    http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
    Good luck with 24 bits.

  • Sample rate for digital sampling (cDAQ-9172 & NI 9401)

    Hi!
    I have a cDAQ-9172 with a NI 9401 C-series module (digital). I would like to sample the digital inputs with a sample rate of e.g. 400 kHz or 200 kHz. My problem is that I can only select a the 100kHzTimebase clock, and therefore only get a 100 kHz sample rate. The 20MHzTimebase clock is too fast, since it gives me a sample rate of 20 MHz). Is it possible to get a user defined sample rate of e.g. 200 kHz, by e.g. dividing down the 20MHzTimebase clock?
    Solved!
    Go to Solution.

    The cDAQ-9172 chassis does not have an internal timing engine for digital input however you can use one of the onboard counters to generate your clock.  Set your pulse train generation counter to be one of the internal counters, such as "cDAQ1/_ctr0" and your digital input sample clock source to be /cDAQ1/Ctr0InternalOutput". 

  • Can't change sample rate for digital input on Mac Pro

    Hello all,
    on my Mac Pro with 10.4.10 I can't change the sample rate for the digital input. Whenever I choose 48000 or 96000 Hz it returns to 44100 Hz after a few seconds. Feeding a 24 bit/96000 Hz signal from an external ADC into the optical input doesn't help. No input signal is available for digital recording software such as Sound Studio or Cubase. After changing the sample rate in audio midi configuration to 96000 Hz the sound can be heard for a few seconds but when the setup returns to 44100 automatically the signal is, of course, lost.
    I've deleted all relevant preferences and restarted with resetting paramter RAM. Still the same. With external hardware such as M-Audio Firewire equipment setting the sample rates works properly.
    Thanks to all for helpful clues.

    Hi,
    when E&M signaling is configured on digital interface like the VWIC is, 2 or 4 wires operation is not applicable because there are no wires at all, and reported only for compatibility with the analog E&M card.
    Consequently, you cannot configure that and it will not make any difference to effect of the connection.
    Please rate post if it helps!

  • How do i ensure my sampling rate is a constant and correct (@250Hz) when using AI Sample Channel vi

    I am running a VI that samples 2 channels using the AI Sample Channel vi within a for loop (executing equal to the number of samples i require), i then calculate the difference between the channels. Within the for loop i use a chart to displays this difference in realtime.
    On exiting the loop the data is converted to an array, which in turn i then convert to a waveform (using dT as 0.01 as an arbitary solution at the moment). The waveform is then compared to limits using the limit specs and limit testing vi's.
    There is also other code displaying graphs from the previous iterations of the VI.
    My question is how do i control the aquisition rat
    e so that i know that the AI Sample Channel vi is sampling my data at a set rate (250Hz)?
    I have tried to use some of the hardware timed exaples supplied by NI to no avail. They can't give me the single point output required within the initial for loop for the real-time display.
    Any solutions welcome!!

    How about buffered acquisition? You can let your DAQCard acquire the number of samples you require instead of doing it with a FOR loop. Using a FOR loop means that you are software timed, which may as well be untimed since it's about as deterministic as the weather in Florida.
    Look for the "Acquire N Scans.vi" example.
    If you want to do this the RIGHT way, use continuous acquisition. Start your acquisition and keep doing AI Read in a loop. Also include your porcessing and display functions in the loop. Just make sure your loop runs fast enough to keep up with your acquisition. At 250 Hz, you should have no problem.
    Dan Press
    www.primetest.com

  • I cant shuffle tracks in playlists like before on latest itunes and when playing ipod through car stereo it will only placks in the 'correct'order - even if i select shuffle on the ipod itself.

    i mostly use my ipod in a car system through the oe radio  (audi a4) - if you connect an ipod and play tracks, evene when setting songs to shuffle it stil plays in the correct order.
    in the past i have made and shuffled playlists then they played fine. since 'upgrading' to the latest itunes it does not now let me do this. you can shuffle tracks but it does not physically re-list the tracks like it used to.
    any help on this woulod be greatly appreciated!
    thanks
    glen

    Read the car user manual or contact the car dealership.

  • Loud sound - the sound is like an incorrect sample rate, a digital input ?

    Hi
    Re: loud sound - I did not explain myself very well. It is a continuous loud hissing sound that pins the meters and sounds like when I record into Pro tools and forget to change from digital to analog inputs. By the way, no trouble recording using Pro Tools LE so I must have some setting incorrect. Thks for your help.
    bye
    Al

    Got an updated Core Audio driver from Digidesign - that fixed it.

  • Bit Depth & Sample Rate: 24 bit 96kHz? 192kHz?

    I am using the Apogee Duet for Mac and iOS on my Mac and I love it - I'm thinking about getting an iPad for mobile recording (voice overs, mostly) and I wonder if Garage Band can manage 24 bit audio at 96 kHz or 192 kHz? I know that the Auria app can, so if nothing else I can just buy that, but since all I would use the iPad for is Voice Overs to edit later in a computer, a $50 app feels like overkill. Comments? Thoughts? Specs?

    Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
    I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
    Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
    Thnx

  • Creative Audigy 2 NX Bit Depth / Sample Rate Prob

    This is my first post to this form
    Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
    The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
    I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
    Here are my system specs:
    HP Pavilion zd 7000
    Intel Pentium 4 3.06 GHz
    GB Ram
    Windows XP Prof. SP 2
    Thnx.
    -cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

    Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
    I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
    Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
    Thnx

  • Maximum audio sample rate and bit depth question

    Anyone worked out what the maximum sample rates and bit depths AppleTV can output are?
    I'm digitising some old LPs and while I suspect I can get away with 48kHz sample rate and 16 bit depth, I'm not sure about 96kHz sample rate or 24bit resolution.
    If I import recordings as AIFFs or WAVs to iTunes it shows the recording parameters in iTunes, but my old Yamaha processor which accepts PCM doesn't show the source data values, though I know it can handle 96kHz 24bit from DVD audio.
    It takes no more time recording at any available sample rates or bit depths, so I might as well maximise an album's recording quality for archiving to DVD/posterity as I only want to do each LP once!
    If AppleTV downsamples however there wouldn't be much point streaming higher rates.
    I wonder how many people out there stream uncompressed audio to AppleTV? With external drives which will hold several hundred uncompressed CD albums is there any good reason not to these days when you are playing back via your hi-fi? (I confess most of my music is in MP3 format just because i haven't got round to ripping again uncompressed for AppleTV).
    No doubt there'll be a deluge of comments saying that recording LPs at high quality settings is a waste of time, but some of us still prefer the sound of vinyl over CD...
    AC

    I guess the answer to this question relies on someone having an external digital amp/decoder/processor that can display the source sample rate and bit depth during playback, together with some suitable 'demo' files.
    AC

  • Sample rate

    I recorded some files with a sample rate of 44100.
    When I went back to cut the files up, I would select the section I wanted to save and "saved selection as" to save the file. I choose the MP3 format.
    Now on the menu I can change the Bit rate and select a mode but I can not alter the original Sample rate. My problem is that Sound booth saved some files at a sample rate of 44100 and some at 48000.
    When imported into Captivate the 48000 files run slow.
    Most likely I will need to install an old version of Audition to batch fix my files. I guess from here on I will use Audition so I don't run into this problem again. Or, since I'm new with Soundbooth one of you will tell me what I screwed up. :|
    Thanks

    The sample rate is generally based on the source media and the mp3 bit rate selection. You could convert the 48K WAV source files to 44K by choosing File, Save As, and selecting the sample rate you wish to use. Then export the converted file to mp3 with your desired bit rate selection.
    I'll need to check and see if this has changed in recent releases, but Flash Player has, in the past, only supported 44K and 22K sample rate mp3 files.

  • What sample rate does the 2 channel oscilloscope example (solution wizards labview 5.1 ) acquire data at?

    I am using labview 5.1 and a DAQCard1200. I chose the Solutions Wizard from the LabView starting window. I followed the steps and selected a DAQ based 2 channel oscilloscope. I chose to program scales etc. myself rather than use the virtual channels. My question is, in this program, when I change the timebase, am I changing the sampling rate? What is the sampling rate for that matter? Is it set to the DAQCard maximum? Does it change if I only set one channel to be active? I threw in some code so that I could take a snapshot of the data, but I end up with 2 columns of voltage data but no idea what the sampling rate was. Makes it very difficult to perform any frequency
    analysis.

    Hi, here's a simple program i did a few years ago. Just remember the daq cards only go up to about 100khz(assuming they have'nt changed) max sampling frequency, so your highest frequency will be 50khz to stop aliasing. This proram will let u select your sampling rate and let u select the number of scans to be aquired. I do not know if the spectrum analyzer side will be any use to u, but its there if u need it.
    Stu
    Attachments:
    op_amp_2.vi ‏104 KB

  • IIR Filtering and response .vi: Butterwort​h filter magnitude response depends on sampling rate -why?

    Hi folks,
    I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
    I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
    Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
    This is fixed and prescribed for the application.
    However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
    To compare the 2nd and 3rd order's magnitude response of the filter I used the example IIR Filtering and response .vi:
    I was very astonished when I the found that the magnitude response is significantly influenced by the SAMPLING RATE I tell the signal generator in this example vi.
    Can you please tell me why - and especially why the 3rd order filter will be worse for the low frequency parts below 1 Hz of the signal. I was told by people experienced with filters that the 3rd oder will distort less the amplitudes which is not at all true for my relevant frequencies below 1 Hz.  
    In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
    THANK YOU for your ANSWERS!!!
    chris
    Solved!
    Go to Solution.
    Attachments:
    butterworth-filter-differences.png ‏285 KB

    Hello Lynn,
    thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
    However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
    I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
    So I assumed that everybody has it in his/her examples shipped with LV and it is not necessary to post it.
    I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
    So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
    I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
    Thanks for any further comments
    chris
    Attachments:
    IIR Filtering and Response_CH.vi ‏55 KB
    butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

  • MIO-16E-1 Output sample rate limited?

    I have a program that I have used successfully on some old Pentium 2 computers. I recently purchased a new PCI-MIO-16E-1 board, and placed it into an AthlonXP 3000+ (2.1 GHz) computer (HP a700n). It seems that I have to limit my output sample rate to be less than 5kHz, where on the old systems the rate could be up to 1MHz. I am wondering if anyone has ran into the problem before or not?
    I have tried the older MIO cards in this newer computer, and came with the same problematic result (no output).
    I have also tried changing the output between IRQ and DMA modes (making sure the input channel is opposite).

    Can you give some additional details of the behavior you are experiencing on the AMD computer? It sounds like if you increase the update rate beyond a certain rate, then your program executes but you don't see a signal on your AO channels. Is this correct?
    Yes, Jeremy, this is correct. It seems that if we go above a 400kHz output sample rate, the signal either doesn't make it to the output buffer, or simply never gets placed on the AO of the board.
    Are you checking for errors in your program? If so, do you get any errors? Are you using NI-DAQmx or Traditional NI-DAQ to communicate with your hardware?
    Yes, we are checking for errors, and we get none. Using Traditional NI-DAQ, all responses are (0) as expected.
    It also sounds like you are performing AO and AI in the same application. Have you tried to run a simple AO example that ships with NI-DAQ?
    I was searching for an example to try when I though about trying a Pentium PC to see if the same thing happened. As far as I know, anytime our software has worked on any computer, I need not go back to trying samples, as they are typically more complex than what we end up with. We simply make a voltage array, and shoot the array to the board (After setting output rate).

  • Digital sample rate

    Is there a way to set the sampling rate for digital inputs?
    In am using an 6032E Daq board. I use two VI's: Port Config and Port Read, but there I cannot set the sampling rate.
    The only thing I found with rate is for handshaking.
    I still have an other question then: in the same VI, I want to read from an Analog input and from a Digital input. I already wrote that VI, and it runs, but is there something I must consider when reading from both analog and digital inputs at a time?

    You cannot set a sampling rate for digital operations on an E-eries DAQ board. The port is static, meaning that there is no timing circuitry controlling the port. It is updated or read by software command only.
    You configure the port once, but you must call port read each time you want read from the port, and port write each time you want to update the port.

Maybe you are looking for

  • CRM_ORDER_DELETE does not delete salesorders with follow-up documents

    Hi All, in my CRM 5.2 i have to delte many sales orders exchanged in past from R/3 to CRM via middleware. There is report CRM_ORDER_DELETE to delete sales document filtering with transacion type and sales order code, and it is ok; but the problem is

  • How to install WD My Book World drive on Yosemite Mac

    I am a novice Mac user and need installation instructions for Western Digital My Book World (model #WD10000H1NC-00) external drive to a Mac (Yosemite 10.10.1). I downloaded and installed WD-Anywhere-Backup-Mac-2.5.125.zip from the WD site, but as am

  • Bw Web Reports

    Hi All, Please can you some one let me know how to add colours for sub totala in web report. Thanks, Sudhakar

  • Tomahawk inputCalendar problem

    Hi i am using the calender tag. The code is given below <%@ taglib uri="http://myfaces.apache.org/tomahawk" prefix="t"%> <t:inputCalendar id="calendar" value="" renderAsPopup="true" popupDateFormat="MM/dd/yyyy" renderPopupButtonAsImage="true" /> The

  • I have used the iPhoto edge blur in aperture but on exporting they look far more blurry, can you help

    I have edited my photos in aperture using the iphoto edge blur tool. On exporting they look more blurry that than they look in aperture why is that? Can I do anything to make my images look the same as when I see them in aperture.