Question audio sampling rate

Similar Questions

• Explore: Discover: Choose Details: Audio sample rate and the size of the Audio sample are not available in the list choose the details.

  With audio files (in particular the WAV), Audio sampling frequency and the size of the Audio sample are not the choices available in the list of details with Vista.  In earlier versions of Windows (2000 and XP) they were both selectable as details.  Is it possible to get these will appear under Vista?

  Vista - related audio details available:
  Album
  Album artist
  Bit depth
  Bitrate
  Duration
  Kind
  Year

  2000 / XP - audio related details are available:
  The album title
  Artist
  Audio sampling rate
  Audio sample size
  Bitrate
  Kind
  Title
  The track number
  Year

  FWIW, sampling frequencies are discussed in the Help window and how to (below).

  Reference: http://windowshelp.microsoft.com/Windows/en-US/Help/53adb4c7-d538-42f8-bb13-917379922afe1033.mspx

  Thank you!

  For the third part of the applications that perform many tasks, I usually discover www.tucows.com and www.download.com.  They have a wide variety of programs, and the trick is to put in the correct search terms to find what you are looking for.  Make sure you that your selection is compatible with Windows Vista and at tucows, try to pick one with 4 or 5 cows because they are the highest rated.

  Good luck! Lorien - a - MCSE/MCSA/network + / A +.

• Question DAQ sample rate

  Hello

  I tried to understand how the 'number of samples' and 'rate' controls affect the frequency of sampling for the DAQ hardware. For example, say I want to acquire data from a sensor of pressure at a frequency of 10 Hz intuitively, I would think everything I do is on the desired sampling frequency, in this case 10 Hz control the 'frequency', try this, I know that's not true. I read that 'number of samples' affects the sample rate by setting a buffer value that must be reached before the VI will process the acquired data. So I also tried to set the "number of samples" to 1 and "rate" at 10, thinking this would have led to a sampling frequency of 10 Hz, and again, it is not. The only way I know to control the sampling frequency is using the wait function (ms), but then I always get buffer overflow errors.

  Can somone if you please explain to me the error in my thought process and also tell me the best way to control the sampling frequency? Is attached a simple VI, I am using to measure my actual sample rate and compare it to the sampling frequency that I am trying to achieve.

  The VI use the DAQ assistant to acquire data of pressure, inserts data into a table, and measure the size of the array. I'm then by dividing the size of the array by the elapsed time in seconds for the sample/s (I'm also dividing the number of iterations of the loop by seconds and using it as a comparison). I compare this value to my entries for the 'number of samples' and controls 'speed' in order to give a sense of the role they play in sampling rate. The VI also allows to choose to use the wait function (ms), as well, using this function is the only way I can control the actual sampling frequency, but then I always get buffer overflow errors. Any information would be helpful, thanks!

  What is the device that you are using? My guess is that whatever you have, it does not allow such a slow pace and is failing at its minimum.

• On the NI PCI-6221 fast sampling rate question

  Hi I was wondering if someone can answer a question of sampling rate on this card to PCI-6221 (http://sine.ni.com/nips/cds/view/p/lang/en/nid/14132).

  Especially if I wanted to transmit simultaneously (analog output) and data acquisition (analog input), what is the sample rate max I could use. Kind regards.

  Since the 6221 is multiplexing the analog input, your question for I / simultaneous ao is possible for one channel of the only. If your "simlutaneously" can include delays (e.g., 100us), you may be able to work with several AI channels as well...

  HAVE the multiplexes, workable sample rate given that the total sample (250 kHz) frequency divided by the number of channels that you use. AO is faster than HAVE it, so it does not reduce this number.

  hope this helps,

  Norbert

• Problem with the audio sampling during export

  I'm having a problem with my audio sampling rate. For some reason, it is set at 8 khz by default when I export. It wasn't a problem until the last update (2 weeks ago). I'll try and give all the info I can to the front. I also tried customer support... which was a nightmare.

  OK, the question that I address is that when exporting audio sampling rate is 8 kHz by default. Which to my knowledge has NO relevant use what so ever. Even when I set the file format to the h.264 format, he insists on the reset to 8 kHz. Even if I manually set all look rendering queue and click on output module again once it back the sampling frequency to 8 khz automatically of changes that I need to set it again. Why can it not only be fixed to 48000 like all other adobe products is and as it was two weeks ago. Does anyone have an answer?

  This problem is resolved by update fixes after effects CC (12.2.1), which is now available:

  http://Adobe.LY/AE_CC_1221

  Note the part at the end of this page on a crucial update for the creative cloud desktop application, which addresses serious problems with the SOUL, first Pro and After Effects.

• How to change resolution/sampling rate a project?

  Is it possible to change the video resolution and audio sampling rates a project first CS5.5? I don't see how once a project is launched.

  A project is not a specific resolution, frame rate and so on. sequences of only. That said, you cannot change the settings of the sequence once it has been created. The workaround is to create a new sequence with the settings that you want, and then copy and paste the content of your original sequence.

• fast sampling rate question...

  Hello

  I use USB-6009 and max sampling rate is about 48 K samples/s according to

  the specification...

  Question 1.

  48 K samples/s means... only when you receive 1 analog input?

  If I have 2 analog inputs then forge would be just half of the 48K?

  Question 2.

  using the daq assistant.

  I would like to get about 50 samples between 10ms

  If I do the math I get 5 K samples/s, which is enough for me

  However, I played with samples to read and throughout the day, the sampling rate,

  do not get this rate... (I'm outputing in file with LVM)

  I searched on the sampling frequency, and people here said

  samples read and sample rate do not havea correlation...

  but I see clearly that they are relevant. When I change a setting

  I get a different number of acquisition... I do N smaples.

  Please help:)

  Q1. Yes, except that the switching of channels takes awhile so the net price per channel is slightly less than half the rate of single channel.  The USB-6009 specification document does not indicate what is the switching time.  You should be able to get 5 kHz on both channels.  20 kHz might be close to the upper limit, but that's just a guess.

  Q2. The DAQ Assistant is often not the best choice for maximum performance.  I do not have the DAQ Assistant, so I can't be more specific. If you get the data as an array of DBL, rather than dynamic data type, it can be recorded directly, without conversion.  The other thing that can make a big difference is a loop two architecture of producer/consumer.  This allows the acquisition of data and save it to the file to run it at different speeds so that each can be optimized separately.  If you are trying to acquire 50 ms of data at a time and then, he writes to the file, you write to the file twenty times per second.  The first time, the operating system must reallocate some file space or do something else what delays write the file, your timing loop is disrupted.

  Lynn

• Audition 3.0 how to disable ASIO and the default Sample Rate recording?

  Hi people,

  New here, but I hope someone can help me with a few questions, I'll have with Audition 3.

  Firstly, some background questions.

  I use hearing parallel to a broadcast audio broadcast program called SpotOn,

  This software requires that I run the sound card, a RME Madiface XT in 48 k mode, and that all the outputs that it uses are defined as WDM Windows so that the windows kernel mixer can control them.

  This means that when I use the hearing at the same time, I have to configure it to use the "Audition 3.0 Windows Audio" driver to stop him from taking control of the sound card directly and change setting which prevent SpotOn to see its output.

  The problems I encounter are that hearing itself seems to randomly change mode in the edit window ASIO driver, I suspect that this happens when I import audio data from a key which is from 44.1 to modify for use in SpotOn. This often seems to not only make the outputs of the card its invisible to the windows kernel mixer but also change the sampling frequency of 44.1 sound card and stops work SpotOn.

  The second question I have is that the sampling frequency of default record when I record in edit mode is always 44.1 and if used it again change the map sound 44.1 and causes the same problems, I'd be very keen to know how to change this default to be 48 k if possible.

  Then.

  What I ultimatly looking is...

  1. a way to disable the ASIO drivers in hearing so that it is not only this option is available, and cannot use the Audition 3.0 Windows Sound Drivers.

  2. a way to make the sampling rate 48 k to stop people choosing 44.1 mistakenly when saving default record.

  Any help or advice that anyone can give would be much appreciated.

  Thanks in advance for your comments

  What other pilots ASIO sees your installation of 3 AA? If it's the Madiface one and you use only WDM drivers you can just uninstall the ASIO RME driver?

• Getting the error «for input and output sample rate does not match...» »

  I am trying to create an audio file, but when I click on the record button I get this error:

  «The sampling rate of the audio output devices and audio don't match.» Audio cannot be recorded until this problem is corrected.

  Use the appropriate operating system or control panel of audio device to adjust the sampling frequency of the output devices and audio entries to use the same settings. »

  I have the preferences set to use operating system controls and it seems to work because when I change in the Win 7 Pro CP it adjusts at the hearing.

  So I do not know which need to be addressed. I use the video card as the source and not the speakers thus mean the output must be la carte too?

  Please help, I'm confused.

  Thanks for your time!

  Got an answer to a previous question asking how to record and thought, this could be useful for others.

  rothmannnyc,

  

  February 18, 2013 16:33 in response to talbot

  It takes a Windows setting and a setting of the hearing.

  You will need to go into your settings from Windows recording devices - if you see a speaker in the lower right of your screen, right-click on it and choose recording devices.  Find your audio input source, select Properties, go to advanced.  You should see a drop down menu with options for the quality and frequency of sampling, mono/stereo.  In general, you'll want to 44.1 kHz for audio; 48 Hz for the video.

  At the hearing, go to Edition/Preferences/Audio Hardware and select the same frequency of sampling for output.  I went through the same question a few months ago, and it worked for me.  Good luck!

• Specific sampling rate?

  So maybe this is a stupid question, but I need to know because I train for a specific sound. Is there a way [to logic] to shoot/change of a certain frequency sampling rates. I can imitate the sound I'm looking for with a low pass filter, reverb and a distortion. But I don't want to 'emulate', this sound, I want to create. Then I can put my own effects and play with it like I want to. If I have to use a bunch of effects to make it sound like I want that also the addition of said effects remove the sound and sound horrible. as to where pulling the sampling frequency of the high frequency and no downs will make me THE noise that I need and always allow to add nice effects to make MY sound instead of someone else. I hope you know what I mean. Let me to you specific real once more. I want to pull or carry a certain frequency sampling rates for a sound under water. I don't want to use filters to make the sound. So can you please help me. I invited everyone locally on how to do it and nothing works. Also if this is not possible in the logic of tell me if there are third party plug ins or maybe even a different DAW that could do like komplete Kontrol or audacity.

  Effect under water

  See if this thread is helpful at all...

• create the audio sample

  I can't find an answer to this question online.  I have a drum track I recorded and want to create a file "sample" of the trap.  The trap played unevenly so I would like to take the best shot of snare and make a sample.  Then I can use this sample with the function "replace or double drum track.  Thanks in advance!

  Odd that you re cannot find how to do or change an audio sample in Logic anywhere because there are many out there if you Google your problem, but here is a simple way to do this:

  Use the scissors to trim the audio file

  Put this file on a new audio track to start at position 1

  Double-click the audio file to open it in the sample Editor

  Adjust the start and end of the snare hit

  Then use the gradient tool, if there is no break at the beginning or end of the audio file.

• myRIO sampling rate

  I'm new to myRIO and use it to measure sine wave (0V to 5V) of up to 10 Hz 20 KHz. I also quickly transformed of Fourier (FFT) of the signal measured in real time.

  Sideways FPGA of things, I try to keep things pretty simple, just read 2 channels of AI (connector B: AI0 and AI1), therefore potentially able to read each HAVE 250 kech. / s (as the unit has a capacity of 500kS/s). Does that mean this program gets a two analog inputs data exactly every 4 microsecond? If this is not the case, how can I make sure that the data is acquired through a fixed sampling rate?

  

  I realized that we can add to the FFT in FPGA function, but I wanted to manipulte the acquired data of analog inputs before it is sent to the FFT, which I don't know how to do now. Can someone explain me how do the arithmetic data (muliplication, division and so) on the acquired data and analog inputs to reducde the 12-bit resolution 10-bit to program FPGAS.

  Later, I created a myRIO program to read analog data 2 FPGA program which continues to turn in timed loop. In the program myRIO, the timed loop is configured to 1 MHz clock source type by a delay of 25 microseconds.

  This configuration means that the loop runs exactly every 25 microsecond?

  When I set up the less than 10 micro second time, myRIO has stopped working. Why is it so?

  Is it because myRIO cannot run as fast as FPGA?

  

  It is advisable to make the FFT of myRIO side analog data or FPGA?

  When I tried to do FFT using the power spectrum of myRIO side, he asked for waveform data. What I acquire is data analog. How can I convert in waveform data?

  If I read in the forum for help, I couldn't have the full answer to my doubts

  Discussions at the Forum I did reference:

  http://forums.NI.com/T5/academic-hardware-products-Elvis/setting-tick-count-in-myRIO-FPGA-software/m...

  http://forums.NI.com/T5/academic-hardware-products-Elvis/myRIO-aggregate-sample-rate/m-p/2707061/HIG...

  A lot of good questions here, I will try to answer as much as I can so as to offer a bit of advice.

  First of all, if you are looking to acquire data at a very specific rate on the FPGA, you'll want to use the Timer VI.  You are also going to use a FIFO of DMA to transfer data of FPGA in real time. A node read-write using as you do now means you'll run out of samples, or read the sample even several times.  The link below is a very good tutorial on how to do what I described above.

  http://www.NI.com/Tutorial/4534/en/

  Later, I created a myRIO program to read analog data 2 FPGA program which continues to turn in timed loop. In the program myRIO, the timed loop is configured to 1 MHz clock source type by a delay of 25 microseconds.

  This configuration means that the loop runs exactly every 25 microsecond?

  When I set up the less than 10 micro second time, myRIO has stopped working. Why is it so?

  Is it because myRIO cannot run as fast as FPGA?

  In general, you should not run a timed loop much faster than 1 kHz.  Using timed inside loop knots, you can monitor the real rate of loop during execution to see if f you meet your needs of the moment.

  The portion of your myRIO RT is slower than an FPGA in the sense where it cannot manage the rates of lines 40 MHz (he makes up for it by being able to work with much better pictures) and it is important to remember that it is just a computer.  The advantage of a real-time operating system, is that you have more control on the Scheduler, not that he is faster (less jitter, not faster code). There is more good reading below.

  http://www.NI.com/white-paper/3938/en/

  It is advisable to make the FFT of myRIO side analog data or FPGA?

  When I tried to do FFT using the power spectrum of myRIO side, he asked for waveform data. What I acquire is data analog. How can I convert in waveform data?

  I would say that it is generally advisable to treat your FFT on the side FPGA as long as you have the resources available, but for many applications probably little matter ultimately.

• DMM (NI 4070), how to correctly set AC Freq (bandwidth) by the sampling rate

  using a NI4070 multimeter and I see the max connection is 300 kHz by respect it.  But I don't understand how to set the min and max, acFrequency according to the sampling frequency or speed reading.

  6 1/2 digits resolution, the speed can vary from 0.25 s/s to 100 s/s and this range corresponds to a lower end on the connection (minimum acFreq) from 1 Hz to 400 Hz.

  (Q1a) - is the playback speed, controlled by the minimum setting of IviDmm_ConfigureACBandwidth?   or vice versa?

  Otherwise, I do not see how to control the rate of reading or the sampling frequency.   IviDmm_ConfigureMeasurement only allows you to control the range and resolution.

  (Q1b) - is there a way to directly control the sample rate (digitizer) or playback speed (dmm)?

  (T2) - the upper limit of the bandwidth of AC always seems to be at 300 kHz... is there still a reason to reduce this maximum value?

  (T3) - Finally, unlike the traditional niDmm function, the resolution via the IVI configuration should be passed as absolute value; does directly when number of digits and the beach?   For example if I want to 6 1/2 digit to 300V range, I guess that by the specifications that the resolution should be set at 0.001 V... followign, if I want 5 1/2 digits to 1V range, the resolution should be set to 0.00001 V?

  Hi Rjohnson,

  I'll try to answer your questions as best as I can:

  Q1A.  The ConfigurACBandwidth function is used by the driver OR DMM to calculate the good aperautre for the measure.  So yes, by adjusting your minimum frequency, you will affect your reading speed.

  Q1B.  Your reading rate will depend largely on your measuring cycle.  To get a fast measuring cycle, there are a few things that you can adjust.  You can programmatically control your time aperature, as well as your time to settle.

  Q2. I can't find a reason to change. This parameter is only used for error-checking and verifies that the value of
  This setting is less than the maximum frequency of the device.

  Q2B.  I think what you say is right, but I'll need to check on that - I'll let know you as soon as.

  Hope that helps.  "" "I would recommend checking the explanation of the Cycle of the DMM measurement in DMM help' devices ' NI 4070" DMM Measuments "DMM measurement Cycle.

  Take care!!

• . VI filtering IIR and response: response of Butterworth filter size depends on sampling rate - why?

  Hi people,

  I'm not an expert in the design of the filter, only a person in applying them, so please can someone help me with an explanation?

  I need to filter signals very infrequent using a buttherwoth filter 2. or 3. order of the bandpass 0.1 to 10 Hz.

  Very relevant amplitudes are BELOW 1 Hz, often less than 0.5 Hz, but there is as well the amplitudes beyond 5 Hz to observe.

  It's fixed and prescribed for the application.

  However, the sampling rate of the measuring system is not prescribed. It may be between say between 30 and 2000 Hz. Depends on the question of whether the same set of data is used for analysis of the higher up to 1000 Hz frequencies on the same measure or this is not done by the user and he chooses a lower sampling rate to reduce the size of files, especially when measuring for longer periods of several weeks.

  To compare the response amplitude of 2nd and 3rd order filter, I used the example of IIR filtering .vi and response:

  I was very surprised when I found that the response of greatness is considerably influenced by the SAMPLING RATE I say the signal generator in this example vi.

  Can you please tell me why - and especially why the filter of order 3 will be worse for the parts of low frequency below 1 Hz signal. Told me of people experienced with filters that the 3rd oder will less distort the amplitudes which does nothing for my the frequencies below 1 Hz.

  In the attached png you see 4 screenshots for 2 or 3 command and sampling rate of 300 or 1000 Hz to show you the answers of variable magnitude without opening labview.

  THANK YOU very much for your ANSWERS!

  Chris

  Hello Cameron and thanks for my lenses of compensation.

  I can now proudly present the solution of my problem.

  It seems to be purely a problem of the visualistion information filters through the cluster of the scale.

  After looking in the front panel of the IIR, I suddenly noticed that the "df" of the pole size is changing with the Fs of the input signal.

  For a Fs to 30 Hz, the "df" is 0.03 Hz so you see the curve of the filter with more points, see png.

  For a Fs 300 Hz "df" is 0.3 Hz, so the curve is larger with only 3 points between 0 and 1 Hz.

  For a 1 kHz Fs the df is 0,976 Hz, so there is no point in the graph between 0 and 1 Hz.

  It's strange that for constant Fs, df of this cluster NOT reduced with the increase in the number of samples, as it does in an FFT.

  However, I hope now the filter used now for the curves obtained with the proposed Lynn way and the response of greatness from the filter information fit together.

  Thank you for your support.

  Merry Christmas and a happy new year to all.

  Chris

• Aggregate sampling rate

  Hello

  I'm looking at possible solutions for data acquisition. I use 4 or 5 entries analog and two digital inputs. During the analogue entered most of them will not need sampling extremely quick rate except for one who needs the least 100ks/s. I noticed solutions cost-effective have overalls sampling rate (eg. 250 ksps / s) which extends on all channels. For these products, such that the NOR-9205 compactRIO module, is possible to distribute unevenly sampling rate between channels (ie. could I give up 100ks/s for a single channel and spread the 150ks/s rest between the remaining channels in use)? Thanks in advance for any help,

  Adam

  Hello Adam,.

  To answer your question on the sharing of the sampling rate, it is not possible to have a single module different sampling frequencies, as described in this KB: here (http://forums.ni.com/t5/Multifunction-DAQ/Aggregate-sampling-rate-and-Multichannel-Scanning-Rate/td-....)

  In the case of the 9205 this module multiplexes between all channels (32 cases set up in single ended mode or 16 in differential) this means that the sampling rate of 250 kech. / s matches total on all channels.

  If you are using the differential mode then the samples per second on each channel will be 250 kech. / s divided by 16, IE 15KS/sec. However if you only specify 4 channels max sampling frequency will be 250 kech. / s divided by 4, IE 62.5 kech. / s.

  One way around this is to use 2 x 9205 in one of our new CompactDAQ chassis, which has 3 engines of timing to HAVE. This allows to set different timings in 3 different modules. What is described in this KB: here (http://digital.ni.com/public.nsf/allkb/E7036C1870F6605686257528007F7A72)

  I'm sorry of this reply took so long, and I hope the above information helps.

  Please do not hesitate to answer questions.

  If you want I could get one of our technical sales engineers to give you a call to discuss further with you data acquisition system?

  Kind regards