Keynote 6.5 Audio/GFX timing issue

I've created a presentation in Keynote 6.22 which has an audio file playing, and a graphic appearing whilst it plays, cued to a specific moment in the audio file (the speakers name).
It works fine on 6.22, but when I run it in 6.5, the audio file stops playing when the graphic appears....
Anyone know of a fix for this?

Reset by deleting the graphic and sound file and replace with originals.

Similar Messages

Audio/Slide Timing Issue in Captivate 4

I have created the following:
A master file in Captivate 4.
Each slide in the master file contains an audio file and a .swf file (also published in Captivate 4).
The master file contains all highlight boxes and audio files.
I have four highlight boxes on a slide in the master file. The corresponding .swf file contains five slides.
Two of the highlight boxes must appear on the first slide of the external .swf.
The next highlight box must appear on the third slide of the external .swf.
The last highlight box must appear on the fourth slide of the external .swf.
When I preview the external .swf project (the five slides) from itself, everything works fine. The audio syncs properly when the highlight boxes appear and the slides change according to their own timing.
However, when I import the .swf into the master file, and run it from there, here is what happens:
The slides in the external .swf do not even all appear. The last one is nowhere to be found. The highlight boxes, of course, appear and disappear according to their properties. But the external slides will not cooperate.
So basically, what I have is audio synced up with the highlight boxes in the master file, but no way to coordinate the syncing of the slides in the external .swf with the audio and highlight boxes in the master file.
Can anyone help?

First thing to check is the frame rate of your master file and external files. They should match otherwise they will be playing at different speeds.
But quite frankly, even if the frame rates DO match, I think you're strategy of having Captivate SWFs playing inside another Captivate movie and expecting everything to sync up properly is doomed to frustration. SWFs inserted into a Captivate movie don't necessarily play as you would expect. The longer the duration of the SWF, the more chance it will be out of sync with the main project.
My honest suggestion is to build all of your slides in the one project. Insert SWFs only for small animations.

IPhone Audio Recording Timing Issues

Hi, I recently recorded a concert and also had someone filming on an iPhone. I'm syncing up the tracks now and I've found that the iPhone gets out of sync every 30 seconds or so. The iPhone recording seems to be a little slower than the actual recording. I can fix it by just splicing the iPhone file every 30s and realigning it, I'm just wondering why this happens. Thanks!

Look at the SignalScope app, I believe they have the specs. Supposedly the regular phone mike is filtered above 8khz, but I think the headset mike/audio in may go up to 44. I think the recorder sample app used 16-bit at 44.1 or 48.

Java Audio Metronome | Timing and Speed Problems

Hi all,
I’m starting to work on a music/metronome application in Java and I’m running into some problems with the timing and speed.
For testing purposes I’m trying to play two sine wave tones at the same time at regular intervals, but instead they play in sync for a few beats and then slightly out of sync for a few beats and then back in sync again for a few beats.
From researching good metronome programming, I found that Thread.sleep() is horrible for timing, so I completely avoided that and went with checking System.nanoTime() to determine when the sounds should play.
I’m using AudioSystem’s SourceDataLine for my audio player and I’m using a thread for each tone that constantly polls System.nanoTime() in order to determine when the sound should play. I create a new SourceDataLine and delete the previous one each time a sound plays, because the volume fluctuates if I leave the line open and keep playing sounds on the same line. I create the player before polling nanoTime() so that the player is already created and all it has to do is play the sound when it is time.
In theory this seemed like a good method for getting each sound to play on time, but it’s not working correctly.
At the moment this is just a simple test in Java, but my goal is to create my app on mobile devices (Android, iOS, Windows Phone, etc)...however my current method isn’t even keeping perfect time on a PC, so I’m worried that certain mobile devices with limited resources will have even more timing problems. I will also be adding more sounds to it to create more complex rhythms, so it needs to be able to handle multiple sounds going simultaneously without sounds lagging.
Another problem I’m having is that the max tempo is controlled by the length of the tone since the tones don’t overlap each other. I tried adding additional threads so that every tone that played would get its own thread...but that really screwed up the timing, so I took it out. I would like to have a way to overlap the previous sound to allow for much higher tempos.
I posted this question on StackOverflow where I got one reply and my response back explains why I went this direction instead of preloading a larger buffer (which is what they recommended). In short, I did try the buffer method first, but I want to also update a “beat counter” visual display and there was no way to know when the hardware was actually playing the sounds from the buffer. I mentioned that on StackOverflow and I also asked a couple more questions regarding the buffer method, but I haven’t received any more responses.
http://stackoverflow.com/questions/24110247/java-audio-metronome-timing-and-speed-problems
Any help getting these timing and speed issues straightened out would be greatly appreciated! Thanks.
Here is my code...
SoundTest.java
import java.awt.*;
import java.awt.event.*;
import javax.swing.*;
import javax.swing.event.*;
import java.io.*;
import javax.sound.sampled.*;
public class SoundTest implements ActionListener {
    static SoundTest soundTest;
    // ENABLE/DISABLE SOUNDS
    boolean playSound1 = true;
    boolean playSound2 = true;
    JFrame mainFrame;
    JPanel mainContent;
    JPanel center;
    JButton buttonPlay;
    int sampleRate = 44100;
    long startTime;
    SourceDataLine line = null;
    int tickLength;
    boolean playing = false;
    SoundElement sound01;
    SoundElement sound02;
    public static void main (String[] args) {
        soundTest = new SoundTest();
        SwingUtilities.invokeLater(new Runnable() { public void run() {
            soundTest.gui_CreateAndShow();
    public void gui_CreateAndShow() {
        gui_FrameAndContentPanel();
        gui_AddContent();
    public void gui_FrameAndContentPanel() {
        mainContent = new JPanel();
        mainContent.setLayout(new BorderLayout());
        mainContent.setPreferredSize(new Dimension(500,500));
        mainContent.setOpaque(true);
        mainFrame = new JFrame("Sound Test");
        mainFrame.setContentPane(mainContent);
        mainFrame.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
        mainFrame.pack();
        mainFrame.setVisible(true);
    public void gui_AddContent() {
        JPanel center = new JPanel();
        center.setOpaque(true);
        buttonPlay = new JButton("PLAY / STOP");
        buttonPlay.setActionCommand("play");
        buttonPlay.addActionListener(this);
        buttonPlay.setPreferredSize(new Dimension(200, 50));
        center.add(buttonPlay);
        mainContent.add(center, BorderLayout.CENTER);
    public void actionPerformed(ActionEvent e) {
        if (!playing) {
            playing = true;
            if (playSound1)
                sound01 = new SoundElement(this, "Sound1", 800, 1);
            if (playSound2)
                sound02 = new SoundElement(this, "Sound2", 1200, 1);
            startTime = System.nanoTime();
            if (playSound1)
                new Thread(sound01).start();
            if (playSound2)
                new Thread(sound02).start();
        else {
            playing = false;
SoundElement.java
import java.io.*;
import javax.sound.sampled.*;
public class SoundElement implements Runnable {
    SoundTest soundTest;
    // TEMPO CHANGE
    // 750000000=80bpm | 300000000=200bpm | 200000000=300bpm
    long nsDelay = 750000000;
    long before;
    long after;
    long diff;
    String name="";
    int clickLength = 4100;
    byte[] audioFile;
    double clickFrequency;
    double subdivision;
    SourceDataLine line = null;
    long audioFilePlay;
    public SoundElement(SoundTest soundTestIn, String nameIn, double clickFrequencyIn, double subdivisionIn){
        soundTest = soundTestIn;
        name = nameIn;
        clickFrequency = clickFrequencyIn;
        subdivision = subdivisionIn;
        generateAudioFile();
    public void generateAudioFile(){
        audioFile = new byte[clickLength * 2];
        double temp;
        short maxSample;
        int p=0;
        for (int i = 0; i < audioFile.length;){
            temp = Math.sin(2 * Math.PI * p++ / (soundTest.sampleRate/clickFrequency));
            maxSample = (short) (temp * Short.MAX_VALUE);
            audioFile[i++] = (byte) (maxSample & 0x00ff);
            audioFile[i++] = (byte) ((maxSample & 0xff00) >>> 8);
    public void run() {
        createPlayer();
        audioFilePlay = soundTest.startTime + nsDelay;
        while (soundTest.playing){
            if (System.nanoTime() >= audioFilePlay){
                play();
                destroyPlayer();
                createPlayer();
                audioFilePlay += nsDelay;
        try { destroyPlayer(); } catch (Exception e) { }
    public void createPlayer(){
        AudioFormat af = new AudioFormat(soundTest.sampleRate, 16, 1, true, false);
        try {
            line = AudioSystem.getSourceDataLine(af);
            line.open(af);
            line.start();
        catch (Exception ex) { ex.printStackTrace(); }
    public void play(){
        line.write(audioFile, 0, audioFile.length);
    public void destroyPlayer(){
        line.drain();
        line.close();

Thanks but you have never posted reply s0lutions before ?? And F 4 is definitely not 10 times faster as stated before I upgraded !!

Audio Video synchronization issues

Hi,
I have a video application that I am building - and
previously I used FMS and TCP connections for video. Now expecting
the latency to be much lesser with UDP, I implemented UDP in my
application and stratus as the RTMFP capable server. I used the
sample video phone application's tutorial do a simple sendStream
and receiveStream for video.
However, unlike the hosted sample application itself, my
audio and video are completely out of synchronization - about 4
secs!! I tried to look at the sample code to see if there were any
special settings that were made. I could not find any.
My questions:
1. Did Adobe make any special tweaks in their hosted sample
application? If so, what are they?
2. Are there any known synchronization issues with using
RTMFP connections for video?
3. Does anyone else also face audio video synchronization
issues?
4. If none of the above is true, what do you think my
mistakes are? Please let me know if I need to add more information
to answer this question.
Thanks in advance.
Rohan

with RTMFP, video and audio can go out of sync if the data
rate of your stream exceeds your network capacity. video is
currently sent with 100% reliability, so once it's queued for
transmission, it's going to be sent (eventually). video is lower
priority than audio, though, so your audio data gets first dibs on
your network capacity. if the video stream's data rate is higher
than what fits in your network (after audio), the video data will
back up until its send buffer is filled, and new camera frames will
stop getting captured. in the steady state, this can look like a
multi-second offset between the video and audio.
try turning your video rate/quality down so that it fits in
your network capacity. video and audio should stay in sync.
with RTMP, there's one network transmission buffer (TCP's)
for all of the parts of your stream (audio & video). when you
have insufficient network bandwidth, the TCP buffer will eventually
fill up and video frames will stop being captured to compensate. so
while audio and video might remain in sync, the total end-to-end
latency will go up. when using RTMFP, the audio and video have
independent transmission buffers, so in cases of insufficient
network resources, the higher-priority audio should remain more
timely but video may fall behind.
-mike

[svn:bz-trunk] 17102: Rewrite all ImageSnapShot tests to avoid any timing issue on various app server .

Revision: 17102
Revision: 17102
Author:   [email protected]
Date:     2010-07-28 11:48:20 -0700 (Wed, 28 Jul 2010)
Log Message:
Rewrite all ImageSnapShot tests to avoid any timing issue on various app server.
Modified Paths:
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureBitmapData.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureImage.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureImageJPEG.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureImageScaleLimitedFalse.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureImageScaleLimitedFalseJPEG.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureImageScaleLimitedTrue.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testCaptureImageScaleLimitedTrueJPEG.mxml
    blazeds/trunk/qa/apps/qa-regress/testsuites/mxunit/tests/remotingService/dataTypes/ImageS napshot/testEncodeImageAsBase64.mxml

I am modifying the correct httpd.conf file on the server, it just doesn't seem to work. - If I put the rewrite rules in the <Directory /> the rewrite works but it adds /Library/WebServer/Documents to the URL.
I also tried putting the rewrite rules in <IfModule mod_rewrite.c> but that did not work either.
mod_rewrite is enabled and running on the server.
I will post the rewrite rules again in the code brackets. Sorry for the long post. - If some one can try them out on their Leopard Server to see if they can get them to work, it would be much appreciated. Again, these work on my Leopard Client but I can't get them to work on Server.
-- The httpd.conf file posted above is just the default conf file found in /private/etc/apache2/
<code>
RewriteEngine On
Options +FollowSymLinks
RewriteRule ^(.+)/$ http://%{HTTP_HOST}$1 [R=301, L]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.php(.*)\ HTTP
RewriteRule (.+)\.php(.*)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.asp(.*)\ HTTP
RewriteRule (.+)\.asp(.*)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.aspx(.*)\ HTTP
RewriteRule (.+)\.aspx(.*)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.htm.(.)\ HTTP
RewriteRule (.+)\.htm.(.)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.cfm(.*)\ HTTP
RewriteRule (.+)\.cfm(.*)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.bak(.*)\ HTTP
RewriteRule (.+)\.bak(.*)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\.inc(.*)\ HTTP
RewriteRule (.+)\.inc(.*)$ $1$2 [R, L]]
RewriteCond %{THE_REQUEST} ^GET\ ([^\?]+)\..(.)\ HTTP
RewriteRule (.+)\..(.)$ $1$2 [R, L]]
<code>

How to fix the Timing issue in Discoverer reports

Hi,
While running the discoverer report in Discoverer plus is taking more than 1 hour to complete( Gen.time + Extract to excel)
where as the same report completes quickly in discoverer desktop.
how to fix the timing issue in discoverer plus
Thanks
Srinivas

Timo Hahn wrote:
There is a problem with autoHeightRows and columnStretching used together in 11gR1.
Have you tried without columnStretching?
Or have you tried if it works in 11gr2?
TimoHi Timo, Thank you very much for taking time to respond to my question.. :)
Back to my question...
I tried removing the columnStretching although my requirement really requires this but no effect really happens even if I remove this.
Based on my investigation on the generated HTML, I notice the following items:
1. A table is being wrapped in a div that is being set at a fixed height.
2. On first load, if your autoheight rows is set to 6, the framework is setting a height of 96px to the div. This height would almost cut the last row of the table.
3. If you try to refresh the page or try to re-PPR the component, the framework resets it to 102px which causes the last row to be fully displayed.
My only concern is that IE is perfectly displaying this while Chrome and FF are having problem.
Based on my understanding, the framework is messing up the height only on first load. Not sure but this is how I see it. I am really not confident also on my findings
and I would most likely hear other's comment.
Thanks.

X-Fi Xtreme Audio & 4GB RAM issue - latest driver?

:X-Fi Xtreme Audio & >4GB RAM issue - latest driver?T Hi All,
Been using an X-Fi Xtreme Audio with 4GB RAM for ages, no issues.
I purchased and installed a zotac 9600 GT which has GB onboard ram, which works fine, but now my X-Fi has started causing general system instability. (processes using heaps of CPU time whilst apparently doing not much, jerky/slow adobe flash, distorted (slow-motion) audio at random times, etc etc.
In the support section here it says:
Audio issue from output from soundcard with computer having 4GB RAM:
<span class="text">
Keywords: <font>4GB ram, installation, xfi, x-fi
Summary: Customers with computers having <font>4GB or above of RAM and are experiencing audio issues, please download the latest driver for your sound card.
However... When this issue first came up I formatted my primary dri've, reinstalled the OS and downloaded the latest drivers for EVERYTHING. The problem has prevailed..
It was only yesterday that I tracked down this problem in the support section here, and tried to verify that my driver is indeed the latest. But device manager lists the driver as "6.0.0.209" when in the support section here the downloads are named ".xx.xxxx" or something. How can I confirm that I have the latest driver? In any case, I'm almost certain that I have the latest, and this issue is still occuring.
I have read in other forums that on other varients of my X-FI card, the latest drivers have not solved this problem either.. sorry I cant be more accurate than that.
For now, I have solved this problem by simply removing 2GB of ram from my PC. Obviously, I'd like to find a solution and put the other 2GB back in!
Thanks in advance for your help!

Could DanielK please mod this as well, so we can try them. Or Daniel, you could you please tell me how you modified the previous drivers from auzentech as I dont know how these CL drivers work and are modified. What is the inf in these drivers, and if you could tell me the deviceID?s of all the x-fi models, or atleast for x-fi xtreme music as it is my modell. Thx in advance!
EDIT:
I have modified the drivers, was not so hard after all. Im uploading the file now, which should have support for all X-FI cards thanks to Daniel's great job with the earlier version.
Upload finished in 4-5minutes. I will post with link soon to rapidshare!

[svn] 4859: -Fix packaging timing issue that was caused by the rebuilding of the air applicationupdater .

Revision: 4859
Author: [email protected]
Date: 2009-02-05 10:15:22 -0800 (Thu, 05 Feb 2009)
Log Message:
-Fix packaging timing issue that was caused by the rebuilding of the air applicationupdater. The recompiled files would get laid down properly only to have the old files put back down on top of them. I also made sure the build directory was removed after updateAIR ran so it would not be included in the package
-removed bundles.properties from the wireframe project
bug:SDK-19128
qa:yes
doc:no
checkintests:pass
Ticket Links:
http://bugs.adobe.com/jira/browse/SDK-19128
Modified Paths:
flex/sdk/branches/i10/build.xml
flex/sdk/branches/i10/frameworks/build.xml
Removed Paths:
flex/sdk/branches/i10/frameworks/projects/wireframe/bundles.properties

Hi Chris,
I have confirmed this is an AIR SDK 13.0.0.83 bug.
I have reported this bug to Adobe Bugbase, and I also included the workaround:
https://bugbase.adobe.com/index.cfm?event=bug&id=3750892
The workaround is:
Simply comment the <externalSwfs> tag in your -app.xml will solve this bug, like this: 
DarkStone
2014-04-28

Flash CS3 Animation and Audio not timed properly when packaged.

I am trying to create a simulation with audio, mouse movement, mouse clicks and screen changes. When I play the movie in Flash, the animation and the audio are timed properly. However, when I package it, the animation falls behind the audio. I am using wavs for the audio - I've tried mp3s also.
I have a screen displayed with a mouse. The audio and a highlight indicate the next step, then the mouse moves and clicks the desired area - with a audio mouse click. The first "click" sound is pretty close to where it is supposed to be, but as the movie plays, it begins to occur earlier and earlier. So much so, that the audio starts to begin prior to animation is complete later in the move. This is not a real long demonstration, only about 1100 frames, but I notice the problem around frame 180.
Any suggestions?

IE stinks. There's a huge performance difference when playing
SWFs in
Firefox and IE. We have seen SWFs reach 100fps in FF 3.0
while the same SWF
taps out at around 40fps in the latest IE browser. It's not
Flash, it is the
browser.
Adobe Certified Expert
www.keyframer.com
www.mudbubble.com
(if you want to email me, don't look)
"AMK4" <[email protected]> wrote in message
news:gfk9ls$6kf$[email protected]..
> This has me baffled to no end. I have a flash file
that's behaving rather
> odd.
> Even the original author doesn't have an answer. I work
on a WinXP system,
> if
> that matters. When I test the movie within Flash CS3, it
works as
> expected: the
> animations ends as the audio fades out. When I publish
it, and I open it
> up in
> Firefox, it works as expected. But that's where it ends.
On the same
> machine,
> if I view it in Internet Explorer, the animation takes
longer - it plays
> slower. Consequently the audio ends before the animation
is done.
>
> I've tried multiple WinXP machines, they all do the same
thing. In Firefox
> it
> works as expected and the same way as previewed within
Flash CS3. But,
> viewing
> it in IE, and the animation plays longer.
>
> Wanna see it for yourself? Look at the original author's
page:
>
http://www.flashmo.com/preview/flashmo_137_intro/
>
> Open it up in IE and Firefox, side by side if you can.
Does it play
> correctly
> in both (animation ends as the music fades) ?
>
> Suggestions anyone?
>

VISA Read timing issues

I am using an RS232 to control an older model Power Supply (OXFORD PS 120-10).
I have successfully written several VI's that all work, the only problem is that VISA Read takes WAY too long. I'm talking 10's of seconds to refresh. I need it have it refreshing in milliseconds or at least tens of seconds for the measurements we need. All of the VI's I have written have the same timing issue.
Attached is the most basic Serial Read/Write VI. Is there any way to improve the Read rate? Or might this just be an instrumentation issue. The strange thing is the Write commands work almost instantaneously (I can seem them on the instruments display).
Please help if you can, I've only been working with LabVIEW for a few weeks and am very must still in the learning process.
Thanks!
Solved!
Go to Solution.
Attachments:
READandWRITE timing test.vi ‏14 KB
READandWRITE timing test.vi ‏14 KB

Do you have the communications protocol for the power supply? If you do not have everything right, you will have problems with communications.
Tens of seconds is a clue that you may be getting timeout errors because the default timeout is 10 seconds. Try placing an inidicator on the error out wire inside the loop (after Read) to see if an error occurs on any iteration. The way you have the VI set up you only see the error on the last iteration of the loop.
You are writing a carriage return to the instrument. If it requires that, it almost certainly sends a carriage return with the response. (This is why I asked about the protocol). If the instrument sends a carriage return (or other termination character), then you should Enable Termination Character on the Configuration VI and set the termination character to the correct value. The default is line feed (hex A or decimal 10). A carriage return is hex D or decimal 13. You must wire the numeric value to the termination character input for any value other than the default. Then change the byte count value (at the Read input) to a number larger than the longest message the instrument will ever send, perhaps 100 or 500. The Read will end as soon as the termination character is received, regardless of the number of characters.
I suspect that this is the problem - the instrument sends fewer than 10 characters in most messages but does send a termination character.
Lynn

MiniDV audio sample rate issue

I cannot get the audio sample rate to match. I have captured miniDV footage (32mhz/12bit). In Sequence settings the audio options for bit are 8, 16, 24. My audio is out of sync!!!! What can I do? I have 2 hours of raw footage and can't proceed! Any help is VERY appreciated. thanks in advance!

Hi again!
Hum ... that´s not as simple as it could be!
That way i´ll have to capture twice?!
There must be another way of doing it , i don´t think Avid can do it and FCP don´t .
As i told you i use both systems and i know they´re limitions (or i think i know!), but it´s strange because Avid Xpress DV 3.5 (from the age of stone ) do it in a blink ... ok ... found an FCP limitation!
Thank you!
If there´s another way ... please feel free to post it!
(and is was supposed to "reply" a "miniDV audio sample rate issue.
HBars

Audio and video issue after Win 10 Upgrade - Split X2 laptop 64-bit

Any solution to the Audio and Video issues on Split X2 13" laptop after Windows 10 FREE Upgrade?? - Intel Core i5 processor- 4GB memory I'm sure others are have the same Audio/video problems: - there is Sound and video playback but the sound and video is repeated slowly forward....impossible to understand the audio. The video is clear but also being played in repeated frames as it moves slowly forward!- sounds coming out of the speakers ok - everything was working great on Windows 8.1 (before upgrading to Windows 10). there must be some kind of driver upgrade for IDT High Definition CODEC BEATS audio... is there a work-around until a real fix is provided?? Regards - VK

I have downloaded and installed the HP recommended IDT Audio Driver using the HP Assistant. The current installed IDT High Definition Audio CODEC driver is: STWRT64.sys 6.10.6498.0 The laptop was rebooted but the problem still the same.....Audio is garbeled and impossible to understand the words...video seems ok when playing Youtube videos. My Split 13 X2 product id is: E2S66AV BIOS F.27

C3120 trunk timing issue on HP B460G1 ?

hello
customer encoured issue of trunking timing issue between
C3120 and HP blade server, unable to get ip from DHCP
server when server is booting up
seems issue resolved in .53SE but i didnt find either
on CCO or HP Web ?
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Did you put the trunk in portfast mode ?
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Geert

Questions about phase difference (possible timing issue) RC circuit

Hello,
Below is the program I am using to measure the phase difference in an RC circuit. Simply put I generate a 2kHz sine wave in LabView and send it to the circuit using an Analog output. Then I measure the output sine wave using an analog output.I also measure this using n oscilliscope. I cna clearly measure the phase difference with the oscilliscope and know it to be approximately 1.4 radians.
Issues with the program:
Different phase difference measured each time the program is run for the circuit. It is also never correct.
Possible causes:
You will notice by looking at the vi I measure the phase from the signal generator. Should I be using a second analog input to measure the sine wave as it is outputted at the start of the circuit?
I mainly think that it is a timing issue. While the phase difference is constant each time the program it varies each on run. So the time each tone measurement begins its first measurement seems to be different each time and causes this different phase reading.
The card I am using is a PCI 6221, is there a timing issue related from switching to input and output acquistion or are they seperate.
Is there anyway to ensure that both tone measurements are measuring phase at the same point in (actual) time?
I would really appreciate any advice or alterations on the program anyone could offer me (I am a college student and LabVIEW is not on our curriculum so I have no support, yet I am using it for my project (D'oh!))
Solved!
Go to Solution.
Attachments:
RC Circuit Test.vi ‏271 KB

I would certainly acquire two signals. Feed the analog output right back into an analog input and then your filtered signals in another.
Initially, I would feed the analog output into both analog inputs and measure the phase delay due to the multiplexed A/D on the card. Once you have that measurement, you can feed in the filtered signal and then measure the phase difference of that signal.
Randall Pursley

Keynote 6.5 Audio/GFX timing issue

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