Bloggie record length

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• The REEL with the right record length

  I've implemented a SQL script to download a 'package' and the definition of 'package body' in a file in a windows environment.
  
  To avoid the lines 'truncate' (divided into two lines) in the file, I use the 'SET LINESIZE v_max_length', where v_max_length is the maximum length of a recording in the package.
  
  But if a record is "a: = 1;", in the file I save "a: = 1; ... ', a record length of 'v_max_length '.
  
  How can I get the right length?
  
  
  Example:
  
  SET LINESIZE 200
  COIL my_package.sql
  
  Select RTRIM (TEXT) from ALL_SOURCE, whose name = 'MY_PACKAGE' and TYPE = 'PACKAGE' by ASC of the LINE order.
  
  SPOOL OFF;
  
  Published by: pacoKAS on 25-mar-2010 07:23
  
  Published by: pacoKAS on 25-mar-2010 07:24

  Why don't you just the linesize 1000 value and
  SET TRIMSPOOL ON

  Then you get the optimal performance for each line

  Concerning
  Marcus

• Sound recorder - record length

  Length of default sound recorder is 60 seconds. How can I change? (Windows XP Home SP3)

  Fortunately, Microsoft has removed the limit of 60 seconds in the version Windows 7 sound recorder. For XP, you save a file of 60 seconds, and then import them, which allows you to have 120 seconds. Then repeat this step to get a file any longer. Microsoft details the steps detailed here: http://support.microsoft.com/kb/82215

• Problems with recording / playback volume and record length

  I turned up the logging level in the in the Audio section on the Control Panel, but that sound recorder (Windows) in entertainment relit at very low volume.  My headset with microphone and speakers work very well for Skype calls, so I think that the microphone is ok.

  Also, is it possible to save more than a minute?

  Thank you

  Hello Tjansen,

  Thank you for your message.  Please, try the following steps:

  Double-click on the taskbar of the speaker icon > click on 'Options' > select 'properties '.

  Make sure that the 'Micro' and 'PC speakers' have check marks.

  In addition, ensure that your output device and Microphone are not silent by displaying the mute boxes on each category

  You can then adjust the volume coming out of your microphone with the volume slider, you can adjust the bass etc.

  To do this, click on 'Options' > select 'advanced controls '.

  With respect to the registration of more than one minute, you can follow the steps below as a workaround, or you can download a 3rd party program that performs the actions you have requested.

  Allow the recorder record silent for 60 seconds.  Click 'Edit' > 'Copy' > click 'Edit' > 'Paste' > 'Insert '.

  This will add 60 more seconds. Click on "Edit" > "Paste" > "Insert" as many times as you need. Save as a file 'model' (name it something like long record.wav)

  Then, when you want to save something for more than 60 seconds, open the long record.wav file and instructions on the existing (virgins).

  Make sure that you save the new record with a new file name.

  Please let us know if it did or did not help to solve your problem.

  See you soon

  Engineer Jason Microsoft Support answers visit our Microsoft answers feedback Forum and let us know what you think.

• How to acquire with NiScope at different sampling frequencies and lengths Records?

  I need to acquire the data of 2 channels of the NI PXI-5114 map two different sampling frequencies high, at the same time. Also, I put 2 different record length. Is this possible?

  I understand that 'Vertical' settings can be configured for individual chains because the function 'Vertical niScope Configure' has 'channels of entry with which we can assign the desired channel. But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Hi AJ_CS,

  Why do you have to be distinct from sampling frequencies on channels separated from the digitizer even?  What different sampling rate do you want?

  But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  You do not have an option to configure the settings of hoirizontal on a channel by channel basis because this concept does not exist in the traditional sense of the use of a scope.  Compatible with the concept of IVI, an oscilloscope traditional benchtop will have only a button or a set of buttons for setting the parameters of synchronization of the unit.  There is therefore no horizontal configuration to separate channels on the scanners NOR.

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Similar to a traditional benchtop oscilloscpe, the device is capable of simultaneous sampling.  But as mentioned above, the channels can not be configured for different sampling frequencies high.

  However, you can ignore data that you think is not relevant.  For example, if you assign 100MS/s CH0 and CH1 to 50 MS/s, then you throw all other samples.

  Alternatively, you can use separate scanners (a channel on each digitizer) and configure them to taste at different rates.  You can set frequencies of sampling on scanners NOR separated and even synchronize them with TClk.

  -Andrew

• Fixed writing the length of records from flat file

  Hi all
  I have a program to write the table data to a file. The sql for this code might look as below:
  
  for r in select * from tablexx partition (partn_nm)
  loop
  UTL_FILE. PUT_LINE (V_FILE, V_BUFFER);
  end loop;
  
  The problem is with the name of patition. This is not a fixed value, and continues to change.
  How can I change the name of the partition on the fly. Something like
  
  v_part_nm: = 'JAN2008;
  for r in select * from tablexx partition (v_part_nm)
  loop
  UTL_FILE. PUT_LINE (V_FILE, V_BUFFER);
  end loop;
  
  Please give suggestions...
  
  Thank you
  
  Published by: user627047 on March 23, 2010 01:16

  user627047 wrote:
  I fight to keep away the comma in the data file. My code is as below

  open SQL_CUR for q'X SELECT rpad(C_GRP_PROD,3)||','|| rpad(N_ACCT,23)||','||rpad(to_char(N_CUST)            from TRMD501_BAK partition( X'||V_DROP_PARTN_NM||')'    ;
  loop
  fetch SQL_CUR into V_BUFFER;
  exit when SQL_CUR%notfound;
  utl_file.put_line(V_FILE, V_BUFFER);
  V_LINES := V_LINES + 1;
  end loop;
  

  Can someone tell me please how to write a record length of fiixed without commas.

  Thank you

  UH... maybe try to take the commas out of it?  :|

   open SQL_CUR for q'X SELECT rpad(C_GRP_PROD,3,' ')||rpad(N_ACCT,23,' ')||rpad(to_char(N_CUST),10,'') from TRMD501_BAK partition( X'||V_DROP_PARTN_NM||')'    ;
                  loop
                          fetch SQL_CUR into V_BUFFER;
                          exit when SQL_CUR%notfound;
                          utl_file.put_line(V_FILE, V_BUFFER);
                          V_LINES := V_LINES + 1;
                  end loop;
  

• Fetch the multi-record acquisitions takes too long

  Hello

  I need to improve the performance of my code in multi record acquisition on the following system.

  LabVIEW 2009, Niscope 3.5.2 map 5122 digitizer on the PXI-1042 with PXI-8106 controller connected to the host PC Dell Optiplex 380 with 4 GB of RAM by crossover cable.

  I tested the speed of transfer of my connection to the chassis using the niScope memory transfer Rate.vi Maximum flow, the result was 110,74 MB/s.

  I need to use the scanner to get the waveform of different seismic sensors and it should be as fast as possible to get different transducers in a reasonable time. I used the Multi files get VI to monitor the rate at which the code executes. I want to acquire 30 signals with 10,000 record length to 100 MECH sampling rate. / s.

  Extraction of 30 records with 10 000 record length takes about 300 ms, that is far too long for my experience. I tried different numbers Records and found that the code takes 10 milliseconds per registration type. I've found that extraction vi takes most of the time. I ran my code both recording Multi search vi. Two sleeves at the rate which is not satisfactory for my application. I appreciate if someone can help me solve the problem.

  Thank you

  Ahmad

  .

  I found that on the pulse generator FRP was set at 100 Hz, which means it sends the impulse evey 1/100 sec or each 10 m I think the program was waiting to acquire the signal. I changed the FRP at 1 K Hz, it became much more quickly.

  Thank you

  Ahmad

• Reading time to fetch multiple records vs. single record

  Hi all

  I have an application where I need to get several records, I use an SMU-5185, and basically, the problem is that it takes too long to retrieve multiple records, I did a test to measure the time it takes for the reading of the hole 32 MB of data and results.

  Parameters:

  Records to Fetch = 8192 (maximum 1 page = 4 KB, 8192 Records = 32 MB)

  Fetch time = 1100 ms

  Records to Fetch = 1

  Samples to Fetch = (2 ^ 25-1500) = 32MB

  Fetch time = 70 ms

  So, I want to know why it takes more time to get the multiplerecords if both are reading the same amount of data? Is there something I can do to decrease the recovery time?

  Thank you

  Hi Rodrigo,

  In fact, I think that the rate of release should be out of the equation, as in the while loop I wait until the end of the acquisition,

  You're absolutely right, I apologize for this oversight.

  I'm you test your code and see several times (about 930ms) Fetch when only pick a channel.  As soon as I specified two channels, the recovery time is an order of magnitude smaller (around 90ms).  Change the sampling frequency, fetch timeout and record length do not seem to affect this disparity in the order of magnitude.

  So to reiterate:

  In the case of single record , changing from one channel to both channels double recovery time.  It's what I expect.

  In the case of multi recording , changing from one channel to both channels decreases the time of recovery by an order of magnitude.  I don't think that.

  Furthermore, when I test in the example shipping niScope EX Acquisition configured standard I do not see this effect.

  (1) can you confirm that, in the case of multi account for your tests, you see that the recovery time reduced when you specify two channels instead of one?

  (2) can you confirm that you are not able to replicate that in the example Acquisition set up ?

  I have tabled a CAR 350194 to address this behavior more with R & D, but unfortunately I can't promise a timeline for the resolution.  I recommend at this stage unless you change Acquisition is configured when you set up your code instead of starting from scratch.  Support at home OR you can call at any time to check the condition of this CAR.

  To answer your question of why a use case takes more time than another, it will take more time to study according to the guidelines of this CAR.  We have already mentioned there is some overhead involved with multi-record fetch, but whether or not the time more or less long you see are good times is something that R & D will have to answer.

  Please let me know if you can confirm what I see by testing.

  -Andrew

• Windows Movie Maker - releasing only audio, is it possible to publish the exact length of the audio and does not include the lot of silent at the end?

  Windows Movie Maker - have 6 minutes of audio spliced, I would like to publish in the form of an audio clip with no video.  I heard that this is possible, and it all works, the 'framework' includes 6 more minutes of space without air circulation that gets compiled with the planned 6 minutes of audio matching 12 minutes of audio rather than the 6 minutes.  Is it possible to MovieMaker 'See' the end of my audio instend, to include 12 minutes in length with it default?  Should I include photos or video for the project to publish only 6 minutes, instead of the 12 minutes?   I hope this makes sense...

  I assume you are using Vista Movie Maker 6 and not Live Movie Maker?

  It is a problem that I've not seen before... the recorded length must be equal
  at times combined clips on the timeline. I guess it's possible
  If you've done a lot of editing/trimming clips... something was corrupted.

  I was wondering... If you import the .wma file saved in Movie Maker and drag
  it to the timeline... it shows the clip as being the length of 12 minutes? If it isn't...
  Perhaps you could he split and right click Delete unwanted 6 minutes and
  then re-save.

  Divide... drag the playback at the end of the music and the type indicator... CTRL + L
  .. .to split the file then right-click of the unwanted part and click on delete.

  Tip: Unlike clips video... audio clips can be dragged to the timeline, so don't forget
  they are dragged all the way to the left to prevent air from dead at the beginning or
  between the clips.

  It may be worth trying to go to... Tools / Options / compatibility... tab and the left click
  the "All default settings" button before the Save.

  And... If the clips audio source are not the. WMA format... it could be a useful
  try to convert them to WMA before you import into Movie Maker.

• When records [i] .isDisabled () returns true?

  Hello

  Please check this code

  ServiceBook sb = ServiceBook.getSB();        net.rim.device.api.servicebook.ServiceRecord[] records = sb               .findRecordsByCid("WPTCP");       String uid = null;
  
        for (int i = 0; i < records.length; i++) {         // Search through all service records to find the         // valid non-Wi-Fi and non-MMS            // WAP 2.0 Gateway Service Record.            ConstantData.data += "\nIs the current record valid?: "                   + records[i].isValid() + "\nIs it Disabled: "                 + records[i].isDisabled();
  

  Is to put this line when used in storm 2 with verizon 3 g enabled

  Is the current record valid?: trueIs it Disabled: false
  
  Data is valid and is enabled
  
  UID at record0: MMS 50009Is the current record valid?: trueIs it Disabled: true
  

  What happens the last line?

  I've found that when no link found

  Records [i] .isDisabled () is true.

  Are there other opportunities in Verizon? Coz, this occurs only with Verizon and not in any other service provider that I used.

  See the documentation for the ServiceRecord.isDisabled ()API:

  "True if this record has no routing information; Otherwise, true.

  So I think it is problem with the routing on your device carrier.

  You can also check getNetworkType() and getType() for more details on this service.

• Maximum length of a message

  We have a UM 405 system tsp802, both subscribers and callers outside when leaving a message for all who hear 30 seconds "you have reached the maximum record length" he then gives the normal send with option 6 options to allow you to add to the message. There is no restriction in the mailbox and we also have external callers setting set to 300 seconds and also length maximum msg on 300 seconds for subscribers. Why we get the prompt and how can we get rid of it. Advanced tool seting also indicates the length maximum msg 120000 s. TIA, Jeff

  Here the most common scenario is that you have only the default active license - by default, the max record length when you have active "demo" license is hard coded for 30 seconds and cannot be bypassed unless you activate a valid license key.

  If this isn't the case, I don't have a lot that I can speculate on with the info here in front of me...

• No record may not be added to the QueueDatabase

  
  It is very simple and basic, but when I add a record to a database with the 'tail' access method, he throws a System.ArgumentNullException. I write a modified test of a "excs_access" code, below:

  /*-
  * See the file LICENSE for redistribution information.
  *
  * Copyright (c) 2009, 2013 Oracle and/or its affiliates.  All rights reserved.
  *
  */
  using System;
  using System.Collections.Generic;
  using System.IO;
  using System.Text;
  using BerkeleyDB.

  namespace Test_QueueDatabase
  {
  Class Program
  {
  public static void Main (string [] args)
  {
  QueueDatabase queueDB;
  QueueDatabaseConfig queueConfig;
  chamois of the chain, dbFileName, keyString;

  Try
  {
  String pwd = Environment.CurrentDirectory;
  pwd = Path.Combine (pwd, "..");
  pwd = Path.Combine (pwd, "..");
  If (IntPtr.Size is 4)
  pwd = Path.Combine (pwd, "Win32");
  on the other
  pwd = Path.Combine (pwd, "x 64");
  #if DEBUG
  pwd = Path.Combine (pwd, "Debug");
  #else
  pwd = Path.Combine (pwd, "Libération");
  #endif
  pwd += ';' + Environment.GetEnvironmentVariable ("PATH");
  Environment.SetEnvironmentVariable ("PATH", pwd);
  }
  catch (System.Exception e)
  {
  Console.WriteLine)
  "Cannot set the PATH environment variable.");
  Console.WriteLine (e.message);
  return;
  }

  Try
  {
  dbFileName = "queue.db";
  If (args. Length > 0)
  dbFileName = args [0];
  }
  catch
  {
  Usage();
  return;
  }

  / * Optiionally remove the existing database file. */
  If (File.Exists (dbFileName))
  {
  While (true)
  {
  Console.Write
  ("{0} already exists.  Delete it? (y/n) ", dbFileName);
  Buff = console. ReadLine (). ToLower();
  If (buff == 'y' | buff == "n")
  break;
  }

  If (buff == 'y')
  {
  Try
  {
  File.Delete (dbFileName);
  }
  catch
  {
  Console.WriteLine ("Unable to delete {0}.", dbFileName);
  return;
  }
  }
  }

  / * Sets up the database. */
  queueConfig = new QueueDatabaseConfig();
  queueConfig.ErrorPrefix = "Test_QueueDatabase";
  queueConfig.Creation = CreatePolicy.IF_NEEDED;
  queueConfig.CacheSize = new CacheInfo (0, 64 * 1024, 1);
  queueConfig.PageSize = 8 * 1024;

  / * Create and open a new database in the file. */
  Try
  {
  queueDB = QueueDatabase.Open (dbFileName, queueConfig);
  }
  catch (System.Exception e)
  {
  Console.WriteLine ("Error opening {0}.", dbFileName);
  Console.WriteLine (e.message);
  return;
  }

  While (true)
  {
  Console.Write ("[empty to leave line] button >");
  KeyString = console. ReadLine ();
  If (chaine_cle == "")
  break;

  Data DatabaseEntry = new DatabaseEntry (Encoding.ASCII.GetBytes (keyString));
  Here, it throws an exception because the key of the record. Data are null (it should be the record number bit array).
  Error in the statement «back BitConverter.ToUInt32 (key.» Data, 0)"which will be called by Append (data).
  queueDB.Append (data);
  }

  While (queueDB.Length > 0)
  {
  KeyValuePair < uint, DatabaseEntry > p = queueDB.Consume (false);
  Console.WriteLine ("{0}: {1}",)
  p.Key, Encoding.ASCII.GetString (p.Value.Data));
  }

  Console.Write ("press any key to exit >");
  Return (true);

  / * Close the slider and the database. */
  queueDB.Close ();
  }

  utility #region
  public static Usage() Sub
  {
  Console.WriteLine)
  ("Use: [database] Test_QueueDatabase");
  }
  utility #endregion

  }
  }

  Hello

  Queue database is used to store fixed-length records, so you must set the record length before opening the queue database, but I didn't know that, in your code. To see the setting of the record length, please visit QueueDatabaseConfig.Length.

  Also QueueDatabase.Length does not tell you the number of records, it tells you just the record length. If you want to know the number of records, see QueueStats.nData.

  Kind regards

  Winter, Oracle Berkeley DB

• How to encode a record Audio of AS3 in PHP

  I am trying to build a way to encode data AS3 audio recording in WAV in PHP. I used a library of WAV in AS3 coders previously, but a record 2 minutes would take 4 to 6 minutes on mobile, depending on the device. So I need to encode to WAV (or any other format that would make usable) in PHP to take the load of the application.

  I have transated this encoder of Thibault Imbert to PHP. See the translation:

  function encode($samples, $channels = 2, $bits = 16, $rate = 44100){
      $data = create($samples);
      $length = strlen($data);
      $bytes = createHeaders($length, $channels, $bits, $rate);
      $bytes .= $data;
      return $bytes;
  }
  
  
  function create($samples) {
      $newBytes = '';
      $bytes = unpack("f*", $samples);
  
      foreach ($bytes as $value) {
          $newBytes .= pack("V*",$value * 0x7FFF);
      }
      return $newBytes;
  }
  
  
  function createHeaders($length, $channels = 2, $bits = 16, $rate = 44100) {
      $RIFF = "RIFF";
      $WAVE = "WAVE";
      $FMT = "fmt ";
      $DATA = "data";
      $header  = '';
      $header .= $RIFF;
      $header .= pack("I*", $length + 44);
      $header .= $WAVE;
      $header .= $FMT;
      $header .= pack("I*", 16);
      $header .= pack("v*", 1);
      $header .= pack("v*", $channels);
      $header .= pack("I*", $rate);
      $header .= pack("I*", ( $rate * $channels * ($bits >> 3)));
      $header .= pack("v*", ($channels * ($bits >> 3)));
      $header .= pack("v*", $bits);
      $header .= $DATA;
      $header .= pack("I*", $length);
      return $header;
  }
  
  

  Which will create a file readable by the right length (if the headers are obviously correct), but there is only silence when it is played. If I change the package size in 'create()' to ' f * ' (instead of unsigned 32-bit little endian int, float), I get static and I can faintly hear the sound in the background. I can't solve this multiplier to clear up the static method, however. (EDIT: before it is mentioned, I know the asterisk is not necessary for most, if not all of the package formats.) It was something that I tried to see if she could fix things and I never removed it)

  The file itself is created as if:

  private function start():void {
            if ( !mic ) {
                      this.mic = Microphone.getMicrophone();
                      this.mic.rate = 44;
                      this.mic.setSilenceLevel( 0, 10000 );
                      this.recording = new ByteArray();
            }
          this.recording.length = 0; //reset byte array in case this is not first recording
            this.mic.addEventListener( SampleDataEvent.SAMPLE_DATA, this.saveSample );
  }
  
  private function saveSample( e:SampleDataEvent ):void {
            while( e.data.bytesAvailable ) {
                      var byte:Number = e.data.readFloat();
                      this.recording.writeFloat( byte );
            }
  }
  
  public function stopRecording():void {
            var file:File = File.applicationStorageDirectory.resolvePath( "audio-recording.wav" );
            this.fs.open( file, FileMode.WRITE );
            this.fs.writeBytes( this.recording );
            this.fs.close();
  }
  

  I'm trying to encode the file is saved in stopRecording(). At this point, I don't know what to do. I tried to translate an other encoder as well, but that seemed to run through the memory as if it was nothing.

  Does anyone have any suggestions as to what is the problem? The two encoders work very well in AS3. Is this a problem with the write file? Problem with writing the audio data in the byte array? Any suggestions will be useful

  So I was unable to find a way to do this via PHP, but I have figure out how do it in a way that is effective and freaking fast in AS3. I realized that the slow fuse of encoding has been a loop in the byte array, read each individual byte, manipulating and write it to another array of bytes. This is an inefficient process, but necessary if you want to ignore the encoder of the recorder. The key was not abstract and rather to encode on the fly, like recording that happens.

  If you record like this:

  private function saveSample( e:SampleDataEvent ):void {
            e.data.position = 0;
            while( e.data.bytesAvailable ) {
                      this.recording.writeShort( e.data.readFloat() * (0x7fff) );
            }
  }
  

  It's the encoding that Thibault Imbert wrote as you do a necessary task (pulling the data from the stream and saving it to a byte array).

  With this, the data is written as it is necessary and sufficient to add the headers to make a WAV file.

  this.recording.position = 0;
  this.recording = this.writeHeaders( this.recording, 1 );
  
  private function writeHeaders( data:ByteArray, channels:int=2, bits:int=16, rate:int=44100 ):ByteArray {
            data.position = 0;
  
            var bytes:ByteArray = new ByteArray();
            bytes.length = 0;
            bytes.endian = Endian.LITTLE_ENDIAN;
  
            bytes.writeUTFBytes( "RIFF" );
            bytes.writeInt( uint( data.length + 44 ) );
            bytes.writeUTFBytes( "WAVE" );
            bytes.writeUTFBytes( "fmt " );
            bytes.writeInt( uint( 16 ) );
            bytes.writeShort( uint( 1 ) );
            bytes.writeShort( channels );
            bytes.writeInt( rate );
            bytes.writeInt( uint( rate * channels * ( bits >> 3 ) ) );
            bytes.writeShort( uint( channels * ( bits >> 3 ) ) );
            bytes.writeShort( bits );
            bytes.writeUTFBytes( "data" );
            bytes.writeInt( data.length );
            bytes.writeBytes( data );
            bytes.position = 0;
            return bytes;
  }
  

  And oila. It's a valid WAV file created without delay after the end of the recording (while I was previously known a delay of 3 times the length of the record on my Nexus 4). I know that's not a very popular topic (Google, Adobe, these forums and StackOverflow were pretty useless to try to solve this problem), but I hope that this solution will help someone along the line.

  EDIT: afterwards, make sure that the ByteArray you save in has its endian Endian.LITTLE_ENDIAN property immediately after instantiation. WAV files require this format and are unreadable if is not set to one.

• The NI PCI-5124 scope with 2 triggers

  Hello

  I would like to ask about the trigger for starting and reference with PCI-5124.

  I found a sample VI on the following link, but it does not work as I expected.

  http://zone.NI.com/DevZone/CDA/EPD/p/ID/2998

  VI 'start_and_reference_trigger.vi' can detect the outbreak starting and reference in my system, and they begin data acquisition.

  He starts data acquisition when both start and reference triggers are entered.

  However, what I would do is:

  1. sampling at 200 MHz with a 1500 with a single channel record length.

  2. receive the signal to start (i.e. 50 Hz)

  3 receive the reference signal (i.e. 50 kHz)

  4. for each trigger reference, I would like to acquire the data, i.e. data of 1500 for each 100 trigger acquisition reference (not with the combination with the shutter release to start)

  Start trigger: _|^|_________________________________________

  Ref   trigger:______|__|__|__|__|__|__|__|__|__|__|__|__|__|__|____

  ^    ^    ^    ^   ^   ^   ^   ^    ^    ^   ^    ^    ^   ^

  hours of relaxing I want to acquire

  Trigger channels I use are:

  Start the trigger: PFI 0

  Trigger for Ref: PFI 1

  With the sample VI 'start_and... '. ", I found that it acquires when the two begin and trigger reference comes and data collection was only after the release of a single reference.

  I hope my explanation is understandable and I can have a solution soon.

  Hi Tom,

  This example configures a trigger starting and reference, but only for a single record. It is inside a loop, so it will continue to require a trigger start and each record. According to your description of the problem, you want to make an acquisition standard multi-record of 100 records, but you want to implement a trigger to start at the beginning of the acquisition. "' To do this, you can simply open the example of shipping"niScope EX Multi Record.vi"found in the example Finder LabVIEW or by browsing in your Start menu programs" National Instruments "NOR-SCOPE ' example. You need to add the property node to define your Trigger source begin to be PFI 0, but other than that, it should work fine. I have advanced and created a simplified with the trigger Start implemented, attached below. I hope this helps!

  

• How to delay a PXI-5122 trigger before routed to string of PFI

  Hello world

  I use a PXI-5122 in a PXI chassis. I want to synchronize with two external devices. The first will send a trigger (with a 10 Hz repetition rate) for PXI-5122. Then PXI will generate a trigger (with a constant delay) in the second.

  It seems that I need to generate a trigger, then export this trigger to PFI 0 line, but I do not know how to delay triggers with a timeframe of 4µs. I read that there is a slight delay between a trigger on the PFI and the first sample. And the length of the cable is also an important factor to consider.

  Could someone give me some suggestions?

  Wednesday,

  Thanks for the drawings, that helps a lot!  Somehow, I see this work (how to set up the scanner):

  1. set up the record length to be 12us (4us trigger samples, 8us after outbreak).  If the sampling frequency is 100 ms/s, that would be a record length of 1200 samples.

  2 configure the position of record reference to 33%.  That's how the digitizer breaks 1200 400 samples according to trigger before triggers and 800 samples.

  3. configuration of triggering immediate reference.  This will allow the acquisition of trigger the moment she gained 400 before triggering samples.

  4. export the "reference trigger (Stop)" to send to Device_2.  This output pulse is of variable width, so if you want consistency, you will need to the Device_2 trigger the rising edge of the pulse, did not not fall m.  Once 400-pre-trigger samples are acquired, this impulse will be sent, and then the scanner will be immediately habitable after initiation of sampling.

  5 configure the trigger of the entrance of Device_1 (10 Hz trigger), as the 'Advance trigger' and 'Start Trigger'.  This will make the digitizer wait this impulse to start sampling before the next record.  We set up, the relaxation of beginning to the 1st record and the trigger in advance for all subsequent records.

  This facility should allow a pretty decent timing, but please test to be sure that it will be sufficient for your application.

  Kind regards

  Nathan