audio samples of voltage <>-?

Similar Questions

• 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.

• 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 audio sampling rate

  I read somewhere on this site that the authors of 3rd party applications have no control over the audio sampling frequency capture. It's all right, because my application can work with the 8 kHz that we are apparently limited to. However, when I leave my app, the phone remained in a State where all the sound elements (Media Player, ringtones, etc.) seems LoFi, as if he's playing at 8 kHz and I need to restart the phone to return to normal. Is this a known bug or I do something wrong?

  It's on a Curve 8900 running 4.6.250.

  Nevermind, I found the problem. There was a race condition in my application that caused destroyApp ends until the capture instance audio player is correctly closed.

• All the 'timecodes' now showing in the form of Audio samples

  It's very strange, timecodes are is not displayed as Audio samples, no temporal codes.  If 00:00:27:12 is displayed as 00:00:27:50000 (or something similar).  How do this back to timecode, googeling tried with no luck. (.

  Click on the key on the monitor of the program and uncheck the option "show Audio time units.

• Problem with audio samples

  Hello

  I made a website with Muse a year ago for a pianist with audio samples that worked for a while, but no more, as you can see here.

  I don't know why since I won't change anything. How can I download new audio files? Where can I find a plug-in? Muse like widgets for video but not audio files...
  Thank you

  Loïc

  Hello

  I checked out the site with complete documentation assets is missing. Then you can try these two things.

  1. connect to the administration panel, go to the Site Manager-> file manager. Create a new folder with the same name, download the files one by one.

  2. create a folder on your desktop and allows you to transfer the folder on the ftp server. To see how works the FTP, check this article

  Catalyst for help | Connect to your site using SFTP

  Let me know if you have any question.

• 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.

• Reading files audio sampled with media player

  Hello.

  On the Simulator, the following code works fine:

  System.out.println("starting con");
  FileConnection con = (FileConnection)Connector.open(loc, Connector.READ);
  System.out.println("making input stream");
  InputStream instream = con.openInputStream();
  System.out.println("Starting player");
  Player p = Manager.createPlayer(instream, type);...
  

  He created the player and later tells the player to begin playing the audio.  The variable "loc" is the location of a file in the audio folder of the sample (ex: file:///store/samples/ringtones/Alarm_Antelope.mid).

  It works perfectly on the Simulator.  When I put it on the device, the try/catch block that surrounds the "InputStream instream = con.openInputStream ();" line is triggered and the exception returns the following error: "net.rim.device.api.system.ControlledAccessException" in the console. A google search has met a person who was trying to record videos, and the suggestion was to start it with the command "invokeLater.  I tried with no luck.

  The only files that play on the device within my program are those who are in the SDCard/blackberry/ringtones folder.  Cannot create flow of entry for any file in the samples directory.

  I manually set all the permissions of the application to allow before running this application.  The application is signed as well.

  So the question is, how I play multimedia files from the examples folder?  Is there an IT strategy that would block this type of access to the files?

  Will give glory to whoever solves this.

  You can't play the tones that distributes the RIM on the device.

  See this post.

• Audio samples of input and output rates do not match.

  If I get an error that supposedly get a lot of people, I think I know the problem of fixing, but I think that it is impossible with the microphone that I use but I want to make sure it's just the microphone that I use and nothing else. So I went in "reading devices" and "recording devices" check if they were both on the same sampling frequency, but I could change the sampling frequency of spekers in properties, but not the MIC... the microphone that I use is the PS2 eye cam, it works very well I had to download "CL eye test" to get it running, but it works , by default the format its deadlock is: 4 channels, 16000 Hz (tape recorder).

  The box is grayed out and I have no idea how to do to change the frequency of sampling for it... so now I can't use the program pretty at all to save

  I am sure that you will not be able to change because it is defined by the hardware built into the microphone. If you want to use with the hearing, you should make sure that windows sounds and hearing are configured for 16 k. Otherwise if there is any software that came with that allows you to record must be able to transfer the audio to hearing to change.

  However, Eye Cam is really a toy. If you are seriously in the audio recording, you'd be better off getting a good microphone and audio interface.

• Change the Audio sampling frequency - post audio recording

  Hello

  Is it possible to change the audio to 2oKHz sampling rate after I taped audio? Is currently 44,10 KHz

  Thank you kindly

  Ali

  I just found this today...

  Under "Publish"-> click on the 'more' (button)->-> "Audio" and finally, select an option under "Bitrate:' '.

  Bitrate (128 Kbps) CD

  Near bitrate (96 kbps) CD

  FM Bitrate (64 Kbps)

  Custom bitrate (slider that starts at 32)

  I changed a course of 68 slides in near CD Bitrate to FM BItrate and size of published file dropped to 50 311 KB 93 288 KB.

• the time display in the audio samples

  I'm trying to select audio and I can choose between something that looks like:

  00:00:00:02:04941

  or

  00:00:00:02.112

  If the second is "milliseconds" it's way way way too coarse and it does not let me enter a 4 digit. If the first samples, which means the 02? Or, how to display the offset in samples?

  Or, if I use hearing, can I select audio and that selection appear first?

  Exhibition of the hearing:

  53175 samples as time

  The timeline panel menu > show Audio time units.

  A first help search for ' time audio units "got me to the next page in less time that it took to create this thread: using Adobe Premiere Pro | Audio editing in the timeline panel

• Pilot hole: speed and height increase after audio dropping during A2DP streamin

  When there is an audio fall (which happens from time to time) while listening to music A2DP, the pilot trying to 'keep up' once the connection has been restored. It does this by pressing the audio so that both audio speed and pitch are increased. So what I mean is: audio interruption, the continuous music, speed and height increasing, after about two seconds the speed and not lower level back to normal.
  It is a very bad audio experience.
  The right thing to do would be to drop audio samples completely which should have been passed in these hundreds of milliseconds rather than keep them.

  Similar experiences?

  BTW, I use headphones JBL REF610.

  Best regards, Kana

  My only works. What Media Player do you use? Your headphones have a function of buffer?

• I can't get the audio to be synchronized with the video editing multicamera?

  Multicamera editing, audio will not be synchronized with the video. I have two clips, 1 Plus 6 iPhone and the other an iPad Mini 2. While the audio wave forms are aligned, they will not match the video... the lips are out of sync. How can I handle this? I use the latest version of FCPX and OS x on an iMac of 2015 end fully charged.

  What are the specifications of media files, the frame rates and audio sampling rates?

• Satellite A110-178: Audio output does not pass the low frequenties

  Hello world

  I just bought a satellite A110-178. However, the audio output of the socket minijack isn't what it should be. Frequencies of less than 200 Hz seem to be switched off. Perhaps the sound card the filter to protect the tiny Laptop speakers, but this is not normal.
  I checked all settings of windows, virtual sound is set to off, and I have reinstalled audio drivers realtek, but still no bass.

  Can someone help me please...

  Hi guys

  Have you heard something called Virtual Sound Toshiba tools?
  I think that it s preinstalled on the A110. On my laptop, this request has been pre-installed, and there are different options that might help clear this issue.

  I put t know exactly what that problem because nobody has described the problem detailed, but you should check the options in open detail settings.
  There, click on the Option button and you will get the new window with the option enable called the audio filter when the audio sample rate is less than 44 kHz

  This you should all play a bit with this good luck tool

• With a sampling of the data with DAQMX, error-200279 occurs when making 2d array dbl

  Hello

  I did a system of simple analog voltage with DAQMX data acquisition.

  It is made for reading of capacitance, where output capacitance value out of a circuit in the periodic voltage signal.

  What I want is to get data from four capacitors simultaneously through four channels, using samples n n (dbl 2d).

  The structure of my VI is almost similar with examples of continuous sampling of voltage in LabView, with the exception of a few other calculations in the loop.

  And for the synchronization of the trigger, I've corrected the edge of release with the external signal from the capacitance reading circuit.

  Version no. 1 has a channel for data input voltage. Version n ° 2 has four channels for the input data.

  While ver.1 can get accurate reading of four capacitors circuit data each (a single channel at a time),.

  ver.2 acquires four channels of data, with a single thin data channel, all the others were wrong.

  I saw a 200279 error occur in the DAQMX read part 2d dbl, so I tried increasing the buffers by changing the sampling frequency or the number of samples, but it wasn't everything.

  I rose for most of the forums with the 200279 error, but the solution would not work on mine.

  Anyone can find the problem? I will attach my screws it may include a bit of Korean language, but most of them are in English, shouldn't be too hard to recognize. Sorry for the inconvenience.

  Oh I forgot, my DAQ is NI USB-6259, and it works in Win XP sp3 and LabView 8.6.

  Thank you.

  Hello Azurenight,

  The 6259 is a DAQ card of the M Series Multiplexed, which means that it is not possible to sample each signal at the same instant, rather the channels are all sent through the ADC even and must be sampled in order. More information on this can be found here:

  LabVIEW Help: Multiplexing compared with simultaneous sampling

  http://zone.NI.com/reference/en-XX/help/370466W-01/mxcncpts/multisimulsamp/

  It may still be possible to get the data you need with the card you have - could you give more information about the maximum eligible period between samples on different channels?

  If you require * real * simultaneous sampling, you will probably need different hardware.

  Kind regards