Stopping Audio Recording before out point crashes?

Using cs5.5 and running everything out and in through a Matrox MX02 LE with MAX
Playing out video and audio from Premiere's Matrox timeline through the matrox box and to an audio booth through hd SDI. audio is coming back through embedded HD SDI through the matrox and into the pc and audio only is being recorded.
If I set an In and Out point and record from one to the other fully the recording will be fine, everything works and I can playback with the recorded tracks under the previously edited footage.
The problem is if the talent messes up and I have to stop mid record and go back the audio is saved and I can save the project but I cannot play anything and when i try to do anything premiere locks up and crashes.
When i restart the program everything is there again and i can resume recording.
This however is a giant pain and not something that should be happening.
I'm wondering if adobe just assumed that all recording would go from in to out and when it hits the out point there is a command prompt that shuts everything down properly but when you stop it manually it just cuts and the program doesnt know quite what happened. Since I can record from the in to out point properly I can't gather any other reason why.
Anyone else run into this problem?

Huh.
Well...I've found Soundbooth far easier to use for this purpose. Maybe try that or Audition, whichever you have.

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Incorrectly stopping audio recording?

Hi I am trying to record microphone audio with the following code and It seem that it is successfully saving a audio file as VLC player can play that file. But it is only the vlc player(which is very error tolerating player!) no other player can play that file!
I think I am not closing/stopping the recording correctly. So Please help me to figure out the problem.
import java.io.File;
import java.util.Vector;
import java.util.logging.Level;
import java.util.logging.Logger;
import javax.media.CaptureDeviceInfo;
import javax.media.CaptureDeviceManager;
import javax.media.ConfigureCompleteEvent;
import javax.media.ControllerEvent;
import javax.media.ControllerListener;
import javax.media.DataSink;
import javax.media.EndOfMediaEvent;
import javax.media.Format;
import javax.media.Manager;
import javax.media.MediaLocator;
import javax.media.PrefetchCompleteEvent;
import javax.media.Processor;
import javax.media.ProcessorModel;
import javax.media.RealizeCompleteEvent;
import javax.media.ResourceUnavailableEvent;
import javax.media.StopByRequestEvent;
import javax.media.format.AudioFormat;
import javax.media.protocol.DataSource;
import javax.media.protocol.FileTypeDescriptor;
* @author Sowrov
public class MicroPhone implements ControllerListener{
    CaptureDeviceInfo microPhoneInfo = null;
    Processor processor;
    Object waitSyn = new Object();
    boolean stateTransitionOK = true;
    public MicroPhone() {
        try {
            //finding a Audio Device
            Vector v = CaptureDeviceManager.getDeviceList(null);
            for (int i = 1; i < v.size(); i++) {
                CaptureDeviceInfo cdi = (CaptureDeviceInfo) v.elementAt(i);
                Format[] fm = cdi.getFormats();
                for (int j = 0; j < fm.length; j++) {
                    if (fm[j] instanceof AudioFormat) {
                        microPhoneInfo = cdi; //take the 1st audio device!
                        break;
                if (microPhoneInfo != null) {
                    break;
            if(microPhoneInfo == null){
                System.out.println("Couldn't find any microphone");
                System.exit(-1);
            //create a dataSource from the audio device
            DataSource mpDataSource =
                    Manager.createDataSource(microPhoneInfo.getLocator());
            //create the processor
            processor = Manager.createProcessor(mpDataSource);
            processor.addControllerListener(this);
            processor.configure();
            waitForState(processor, Processor.Configured);
            processor.realize();
            waitForState(processor, Processor.Realized);
            System.out.println("Audio Processor is ready");
        } catch (Exception ex) {
            Logger.getLogger(MicroPhone.class.getName()).log(Level.SEVERE, null, ex);
    public Processor getProcessor(){
        return this.processor;
    private void waitForState(Processor p, int state) throws Exception {
        synchronized (waitSyn) {
            while (p.getState() < state && stateTransitionOK) {
                waitSyn.wait();
        if(!stateTransitionOK){
            throw new Exception("Couldn't Change the sate to: "+state);
    public void controllerUpdate(ControllerEvent evt) {
        System.out.println("MicroPhone Event: " + evt.getClass());
        if (evt instanceof ConfigureCompleteEvent ||
                evt instanceof RealizeCompleteEvent ||
                evt instanceof PrefetchCompleteEvent) {
            synchronized (waitSyn) {
                stateTransitionOK = true;
                waitSyn.notifyAll();
        } else if (evt instanceof ResourceUnavailableEvent) {
            synchronized (waitSyn) {
                stateTransitionOK = false;
                waitSyn.notifyAll();
        } else if (evt instanceof EndOfMediaEvent||
                evt instanceof StopByRequestEvent) {
            evt.getSourceController().stop();
            evt.getSourceController().close();
    public static void main(String arg[]) throws Exception{
        //initializing Microphone setting
        MicroPhone mp = new MicroPhone();
        //getting the processor
        Processor p = mp.getProcessor();
        //getting the dataSource
        DataSource ds = p.getDataOutput();
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                                       ds,
                                       null,
                                       new FileTypeDescriptor(FileTypeDescriptor.WAVE));
        //getting realized output Porcessor from the output ProdessorModel
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        DataSource outDs = outputProc.getDataOutput();
        String outputFile = "file:" + System.getProperty("user.dir") +
                    File.separator + "audio.wav";
        //creating output MediaLocator
        MediaLocator outputLocator = new MediaLocator(outputFile);
        //getting the DataSink for output
        DataSink outputDataSink = Manager.createDataSink(outDs, outputLocator);
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        p.start();
        //start datasoruce
        ds.start();
        outputDataSink.open();
        outputDataSink.start();
        outputProc.start();
        int time = 10;
        while(time-->0){
            Thread.sleep(1000);
            System.out.print(".");
        if (outputDataSink != null) {
            outputDataSink.stop();
            outputDataSink.close();
        //ds.stop();
        //p.stop();
        //stoping input datasource and processor doesn't make any difference
        p.close();
        //javax.media.
        if (outputProc != null) {
            outputProc.stop();
            outputProc.close();
        System.err.println("Done!");
}

The correct order for a processor / datasink is as follows...
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Dear All,
I have just purchased a Sony ICD-UX70/UX80 digital audio recorder to use as a back-up emergency audio source and it works very well in capturing stereo digital audio. However, when trying to edit this audio source into my video, the audio keeps "drifting," I assume because of the difference in the frames per second between video and audio. Is there any "quick fix" to keep both audio sources in sync...a "make fit" that will adjust the audio to the space allotted to the video? Any suggestions?
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Hello,
I have three sources of media that I'm using in my project, which is a recording of a gig that my band played:
- Sony AVCHD video camera (front view), includes one video and a stereo pair
- Sony AVCHD video camera (side view), includes one video and a stereo pair
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However, within 30 seconds before or after the sync point, some drift is evident in the audio that was captured on the R16. And a few minutes out, it's seconds off. I capture these gigs one set at a time, so we're talking about clips that are an hour long (we usually do two or three sets).
I didn't pay much attention to the detailed settings of the audio recorder before I captured the gig. Actually, I recall that the basic marketing/quickstart tutorials on the Apple website say you can pretty much use any video or audio in a single project in FCE and it would convert/transcode for you as necessary.
Had a look at the clip properties, and I see that the audio captured in the cameras is 48.0 KHz, while the R16 captured the audio at 44.1 KHz. If I'd noticed the difference beforehand I would have changed the R16's settings to match, just to be thorough - but I am a little disappointed - I'd thought that there would be compensation for differences like these... this is digital!
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fudster wrote:
RyanManUtd wrote:
Hmmm ... honestly .... I am running out of ideas ...
Can I ask:
1. did you get both their starting point correct ?
2. Are the duration of both the video clips & audio clips you are trying to sync the same length ? That is, if the audio is 10 secs, then is the video also 10 secs ?
Thanks
Hey -
1. Yes, I'm certain that I matched up the in points correctly while making the edits. In the audio tracks, I zoomed in all the way so I could get sub-frame position - as described starting on pg 474 of the manual. At that point, they are in perfect sync. And as you get further away from that matched point in the sequence, the drift gets worse.
2. The length of the audio that was recorded on the cameras is exactly the same length as the video (of course). The audio clips recorded on the R16 are all exactly the same length as each other as well.
However the R16 clips are not the same length as the video clips, and the two videos are not the same length as each other. They are a few seconds different from each other, because of the way I start the recording - I step onto the stage and before we start playing, I use the infrared remote to start the recording on each of the cameras one at a time, and then I start the recording on the R16.
I don't actually know how exact clip lengths from separate recording sources could be achieved, and since I'm editing them in separately, I think it shouldn't make a different. The keyword being "shouldn't", of course. In any case, if you extend the clips all the way out to play all the available media in the track (using up all the handles), the extents of the clips in the tracks appear a few seconds different from each other, as they should.
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bobhugejunk

YOU ARE THE MAN!!!! It took a little bit of working the configurations, but I got it to work....oh, me so happy...oh, oh, me so happy
Thank you!
Make sure you visit www.bobhughesmichigan.com for my music samples, schedule and video information!
Date: Wed, 21 Oct 2009 19:34:20 -0600
From: [email protected]
To: [email protected]
Subject: Problems with 1.5 and vista - audio recording
As Jim points out, CS3 was the first Vista-certified version, though many did get PrPro 2.0 to run on it.
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>

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I'm editing a documentary on Final Cut Pro 5.0.
Using a SONY DSR-11 firewire deck.
I logged sets of clips and then batch captured them. I DIDN'T select add handles in the batch capture tool. Some clips, as they're captured, change their in and out points. Sometimes by two frames early, sometimes by two frames late, and sometimes not at all. Several clips even stopped captured with an out point twenty seconds earlier than what I'd logged. The result is that I have clips with media start and media end points that are not in what I wanted. There seems to be no rhyme or reason whatsoever to this.
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Check to see if your tapes have NDF time code.
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When Using Adobe Premier Pro a few years back I seem to remember playing a tape all the way through, marking In/Out points, all the way along the tape, then going back to the beginning click Batch Capture ( or similar) and the machine would rewind the tape and run through again capturing just the relevant clips.
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Message was edited by: Neil Paisnel

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When I rebooted, it would not allow me to log in under any name, as no user existed anymore. Eventually loged back in from terminal/Single user mode, can t remember now. Finally got back in but dock icons all gone, replaced by big question marks, had to run everything from terminal window..
What I am getting at is that I had been lead to believe that this OS was so solid, and to quote one of my friends ' you cant do anything wrong...it wont let you' I had wanted to believe this though I did doubt it. I use a varity of systems, FreeBSD/FreeNAs server or three, Ubuntu variants etc.
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Sliding audio IN/OUT point in smaller increments than a single frame.

Hello there...
I saw this work-flow on the forum a while ago, and i've searched the archives but can't remember what the original thread was about/titled...
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I have one word I really can't get a clean cut on. The playhead rests just before or just after the moment the next word comes in... there IS a tiny gap on the audio waveform... i just can't get the playhead on it...
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Shane (or anyone else!)... can you advise me?
Message was edited by: James M.

the info you're after is contained in the thead here
http://discussions.apple.com/thread.jspa?messageID=4740000&#4740000
but i don;t think you want to really slip the audio unless it is truly out of sync with the video ... in your case you probably just want to zoom right in on the timeline and edit the audio level to mute as required. audio keyframes can be placed and edited at up to 1/100th frame intervals

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I am having to save the project after each recording, on the chance the next one will crash the program. This is taking forever. None of the projects I'm working on are over 75 slides, and they are between 8-12Mb in size.
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