Bounced WAV files's timestamp doesn't match timeline issue.

I'm mixing down some music for picture. In Logic Pro 8.01, the audio/video sync perfectly. But the bounced WAV files' timestamp does not match. For example, in Logic Pro/Video's timeline, it is starting at 01:00:00:00, but the resulting WAV file's timestamp is at 01:00:03:18. Or Logic's timeline start at 01:13:22:00, the resulting WAV file's timestamp is at 01:13:26:12. I checked it with Pro Tools and BWAVReader.
Logic is set at 29.97 Non Drop and 48kHz matching the DV file. Am I doing something wrong or is this a bug? Any help would be appreciated. Thanks

my preference would be to get Audition 3 (too bad we can't get an upgrade price from CS3 Production Premium to Audition 3...).
Are you sure. I got Audition 3 for US$99, IIRC, from CS2 Production Studio (contained Au 2.0). That was just after the release of Au 3, but not THAT long ago. With CS3 Production Premium, I would think that you would qualify even more than I did.
One point - I'm not using Reverb as an effect...
Then that was my error. I thought that you HAD added a bit of Reverb - sorry. And no, I was talking about the Effect, not the room ambience. Just a mistake on my part.
I am going to try a work around tonight... I am going to export each of the 2 audio tracks separately (with effects applied) and then bring those 2 WAVs back into PP3 to be mixed and just drop the original audio to see what happens
I often do similar to what you are doing - Upmixing. I know that the music industry is opposed to this with re-releases of music, where original multi-channel masters do not exist, but for my work, it is OK. I'll use duplicate Audio, just as you are, and then mix each 2-channel Clip a bit differently, using a touch of EQ, Reverb and Delay (most often), plus the spatial location of the sources for DD 5.1 SS mixing. Now, I am keeping the Effects to a minimum, but the Audio is room-filling, when I do it right, and with a light-hand. For my sources, I have found subtle is better, unless I need a panned SFX, or similar, and then I get to go a bit wild, like landing the helicopter on the back of the audience's heads!
Have you tried SoundBooth for the Audio mixing? I only used it in Beta, but had Audition, so never saw the use for it in my workflow. Who knew that Audition would soon be only a stand-alone?
Good luck, and please let me know what you find out with your tests. I can always learn more about the Audio-side of the street.
Hunt

Similar Messages

  • Help - canvas doesn't match timeline -- ghost images appear

    Hi,
    After having no problem in panther and final cut 5.0 I've had to upgrade to tiger (10.4.8) and FCP 5.1.2. When I opened and edited my project, I got a sort of ghost image (a still image from elsewhere in the timeline) appearing in the canvas over where another clip should be. This happened once and I tried everything (deleting and then brining in clean copy of clip) and no dice. The good news is that when I output to QT.mov, everything looks as it should.
    But suddenly, I'm having this problelm in many areas of my project, which makes it hard to edit at all. Has anyone else had this problem? I've tried rendering everything, but it doesn't change anything.
    Two other clues:
    1) at the same time I've had trouble getting VoiceOver to recognize where i am in the timeline so have had to create "In" marker for the project at that spot where I'm doing a new voiceover.
    2) For what it is worth, there does seem to be some odd lines (same color as rendered project but as if cut into smaller pieces in the area of the timeline that triggers this).
    please help! (and thanks in advance)

    Glad to help. I'm glad to see someone else is working in NY on this cold, gray day. One thing I cannot recommend too highly; browse this forum every day. You see a topic that might be useful, read it. Pretty soon, you'll be an expert at troubleshooting fcp.

  • Why, when I bounce to iTunes, do my wave files show up with no sound?

    When I bounce wave files to iTunes the file has a time and the title shows up, but no sound when you double click on it. The time bar moves like its playing but no sound!!

    If you bounce as AAC or MP3, you will lose some quality. Better to bounce in AIF or WAV if you intend to burn to CD. Now, why you can't play it in iTunes or from finder, that is weird. Like someone suggested, try AIF format. That is all I use and I can play anywhere. When you go to "Get Info" in iTunes (for the song), does it have a time? You can also try opening the file (from your Bounced folder) into Audacity? It's a free app and very nice. You will be able to see the sound wave. This will verify that you actually have sound in the file.
    Let us know what you find.

  • HI, I am having trouble when importing .wav files into logic.

    Everytime I import this .wav file of a DJ set i recorded, it shows me the waveform but no sound comes out when its played. I have tried converting the bitrate to 16bit and the filetype to aiff as I saw it had helped people with similar problems but when i do this its boosts the gain massivly and the recrding becomes very distored? Really dont know what to try??
    Also the reason I am trying to get the Wav in was bevause i need to normalise it as it had to be recorded very quitely to avoid distortion on the recording. I recorded it by using the imput on the back of my monitors as they are usb, and th programme i used to record it is "audio recorder tool" from the app store
    Thanks for your help, Joe

    I think the problem is you're recorded 32-bit wav files which Logic doesn't support.
    (don't convert to aiff, unless you have the right program, that understands converting little endian 32 bit wavs to big endian aiff format, you will get distortion)
    I think your best bet is to download "Audacity" the free audio editor and convert to 24-bit WAV files. Re-save the audio file under a different name after converting. In Audacity this is done using the Export function, selecting other non-compressed formats.
    http://audacity.sourceforge.net/download/mac
    The reason you want 24-bit WAVs to work with is you're going to be raising the volume of the file, 24-bit files will suffer less damage when editing with plugins, normalizing...etc.

  • "This wav file could not be read"

    I have a DL project with 30 timelines. It has been playing and burning fine for the last 4 months--I've just been tweaking menus and replacing some of the assets with newer encodes.
    Yesterday when I attempted to preview I got the message "this wav file could not be read." There are 35 .wav files in the project. This morning I did a "save as" on the project and systematically removed the wav files from the timelines two at a time until I found what was causing the problem (naturally it was the very last one). This was not a wav file that had been replaced in recent months, it has been in the project from the beginning
    Then I opened the original project and deleted only that .wav file from the timeline and the project previewed fine. So I deleted the .wav file from the project and then imported the same .wav file again, placed it in the timeline and the project works fine.
    Can someone tell me what would cause this behavior? Is there some sort of corruption in the asset and I'm going to have the same problem again? Or is it more likely some corruption in the project, and if so what could it be or what should I do about it?

    I have just replaced all WAV files in the project with ac3 files("reaplace asset" command).
    Still, when building DVD, god damned Encore CS4 mumbles "this WAV file could not be read."
    WTF?
    I looked through several topics on the Adobe forums where people ask the same question. There is no one a single answer! So, I'm back to Sonic Scenarist!
    No time for "dancing with a tambourine," around a blunt Adobe.
    And by the way, when Encore telling "This wav file could not be read" -- WHY it's not telling me WHICH ONE file? Do I have to guess or ask a fortune-teller?

  • Playing a wav file (byte array) using JMF

    Hi,
    I want to play a wav file in form of a byte array using JMF. I have 2 classes, MyDataSource and MyPullBufferStream. MyDataSource class is inherited from PullStreamDataSource, and MyPullBufferStream is derived from PullBufferStream. When I run the following piece of code, I got an error saying "EXCEPTION_ACCESS_VIOLATION (0xc0000005) at pc=0x7c9108b2, pid=3800, tid=1111". Any idea what might be the problem? Thanks.
    File file = new File(filename);
    byte[] data = FileUtils.readFileToByteArray(file);
    MyDataSource ds = new MyDataSource(data);
    ds.connect();
    try
        player = Manager.createPlayer(ds);
    catch (NoPlayerException e)
        e.printStackTrace();
    if (player != null)
         this.filename = filename;
         JMFrame jmframe = new JMFrame(player, filename);
        desktop.add(jmframe);
    import java.io.IOException;
    import javax.media.Time;
    import javax.media.protocol.PullBufferDataSource;
    import javax.media.protocol.PullBufferStream;
    public class MyDataSource extends PullBufferDataSource
        protected Object[] controls = new Object[0];
        protected boolean started = false;
        protected String contentType = "raw";
        protected boolean connected = false;
        protected Time duration = DURATION_UNKNOWN;
        protected PullBufferStream[] streams = null;
        protected PullBufferStream stream = null;
        protected final byte[] data;
        public MyDataSource(final byte[] data)
            this.data = data;
        public String getContentType()
            if (!connected)
                System.err.println("Error: DataSource not connected");
                return null;
            return contentType;
        public void connect() throws IOException
            if (connected)
                return;
            stream = new MyPullBufferStream(data);
            streams = new MyPullBufferStream[1];
            streams[0] = this.stream;
            connected = true;
        public void disconnect()
            try
                if (started)
                    stop();
            catch (IOException e)
            connected = false;
        public void start() throws IOException
            // we need to throw error if connect() has not been called
            if (!connected)
                throw new java.lang.Error(
                        "DataSource must be connected before it can be started");
            if (started)
                return;
            started = true;
        public void stop() throws IOException
            if (!connected || !started)
                return;
            started = false;
        public Object[] getControls()
            return controls;
        public Object getControl(String controlType)
            try
                Class cls = Class.forName(controlType);
                Object cs[] = getControls();
                for (int i = 0; i < cs.length; i++)
                    if (cls.isInstance(cs))
    return cs[i];
    return null;
    catch (Exception e)
    // no such controlType or such control
    return null;
    public Time getDuration()
    return duration;
    public PullBufferStream[] getStreams()
    if (streams == null)
    streams = new MyPullBufferStream[1];
    stream = streams[0] = new MyPullBufferStream(data);
    return streams;
    import java.io.ByteArrayInputStream;
    import java.io.IOException;
    import javax.media.Buffer;
    import javax.media.Control;
    import javax.media.Format;
    import javax.media.format.AudioFormat;
    import javax.media.protocol.ContentDescriptor;
    import javax.media.protocol.PullBufferStream;
    public class MyPullBufferStream implements PullBufferStream
    private static final int BLOCK_SIZE = 500;
    protected final ContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW);
    protected AudioFormat audioFormat = new AudioFormat(AudioFormat.GSM_MS, 8000.0, 8, 1,
    Format.NOT_SPECIFIED, AudioFormat.SIGNED, 8, Format.NOT_SPECIFIED,
    Format.byteArray);
    private int seqNo = 0;
    private final byte[] data;
    private final ByteArrayInputStream bais;
    protected Control[] controls = new Control[0];
    public MyPullBufferStream(final byte[] data)
    this.data = data;
    bais = new ByteArrayInputStream(data);
    public Format getFormat()
    return audioFormat;
    public void read(Buffer buffer) throws IOException
    synchronized (this)
    Object outdata = buffer.getData();
    if (outdata == null || !(outdata.getClass() == Format.byteArray)
    || ((byte[]) outdata).length < BLOCK_SIZE)
    outdata = new byte[BLOCK_SIZE];
    buffer.setData(outdata);
    byte[] data = (byte[])buffer.getData();
    int bytes = bais.read(data);
    buffer.setData(data);
    buffer.setFormat(audioFormat);
    buffer.setTimeStamp(System.currentTimeMillis());
    buffer.setSequenceNumber(seqNo);
    buffer.setLength(BLOCK_SIZE);
    buffer.setFlags(0);
    buffer.setHeader(null);
    seqNo++;
    public boolean willReadBlock()
    return bais.available() > 0;
    public boolean endOfStream()
    return willReadBlock();
    public ContentDescriptor getContentDescriptor()
    return cd;
    public long getContentLength()
    return (long)data.length;
    public Object getControl(String controlType)
    try
    Class cls = Class.forName(controlType);
    Object cs[] = getControls();
    for (int i = 0; i < cs.length; i++)
    if (cls.isInstance(cs[i]))
    return cs[i];
    return null;
    catch (Exception e)
    // no such controlType or such control
    return null;
    public Object[] getControls()
    return controls;

    Here's some additional information. After making the following changes to MyPullBufferStream class, I can play a wav file with gsm-ms encoding with one issue: the wav file is played many times faster.
    protected AudioFormat audioFormat = new AudioFormat(AudioFormat.GSM, 8000.0, 8, 1,
                Format.NOT_SPECIFIED, AudioFormat.SIGNED, 8, Format.NOT_SPECIFIED,
                Format.byteArray);
    // put the entire byte array into the buffer in one shot instead of
    // giving a portion of it multiple times
    public void read(Buffer buffer) throws IOException
            synchronized (this)
                Object outdata = buffer.getData();
                if (outdata == null || !(outdata.getClass() == Format.byteArray)
                        || ((byte[]) outdata).length < BLOCK_SIZE)
                    outdata = new byte[BLOCK_SIZE];
                    buffer.setData(outdata);
                buffer.setLength(this.data.length);
                buffer.setOffset(0);
                buffer.setFormat(audioFormat);
                buffer.setData(this.data);
                seqNo++;
        }

  • Export wav file with matching dropframe timecode

    I'm sure I'm just missing something simple, but I need to export a small corrected part of a finished as a WAV file with timecode that matches the project. Being for broadcast in the US, the project is in dropframe timecode. While I'm working in STP, I see drop frame time code. But when I export, I get non-drop timecode that doesn't match the orginal code. This is a five second fix for a one hour DVD and I hate to have to send the authorers an entire new sound track when they only need the five seconds with matching timecode.
    How can I export and make this work. Without having to get ProTools, that is!

    If you're talking about timecode stamping in a WAV file exported from STP . . . well it doesn't seem to work for me no matter what timecode rate I use. 29DF, 29, 24, 23. I suspect you need to either give them the timecode location manually or export the whole thing, and let them just punch in the section they need.

  • I have been buying apps for a long time  without a problem using my credit card on file. All of a sudden itunes is not recognizing my card which is in good standing. All the info is correct but itunes/apple says it doesn't  match the bank info. Any ideas

    I have been buying apps for a long time  without a problem using my credit card on file. All of a sudden itunes is not recognizing my card which is in good standing. All the info is correct but itunes/apple says it doesn't  match the bank info. Any ideas?

    Answered in your Other post on this Topic...
    https://discussions.apple.com/message/24053626#24053626

  • Bouncing as WAV file

    I've got my trusty manual out, and it's telling me that in order to bounce as a .wav file (as opposed to .mp3) I should go to the "Bounce Dialog Window" and select PCM format (which I guess is SDII, AIFF or WAV and thus what I want).
    Problem is: I have absolutely no idea where the Bounce Dialog Window is located. When I go to FILE > BOUNCE (or just click "Bnce" on my output in the Environment Window) a Save As window pops up, wherein I cannot see any option to change from MP3 to PCM format. That can't be it. Or if it is, perhaps I've disabled something important somewhere?
    I've now opened every single window I can think of to look for this and am stumped. Any help would be most appreciated. Danke.

    Thanks Walter.
    Unfortunately, the same issue happens... In whichever window I select "Bounce" from -- Environment, Audio Mixer, or even FILE > BOUNCE -- a window titled "Bounce 'Output 1-2' " pops up. I'm guessing this is what you mean by the "Bounce Output Window." But, there I'm only able to name the file in the "Save As" field, choose where to bounce the file to on my harddrive, and then click on "Bounce" in the bottom-right of the window. I have no tabs or anything else to choose from. So I'm still a little stumped. Any additional thoughts?
    I really appreciate the help.

  • Saved file doesn%27t match original text

    Using LV 2010.  I have an application where I'm encrypting a file and saving it to the hard drive as a text file.  The problem I'm having is that the string of characters that I'm writing to the file does not match the string of characters that are read from the file.  The files match for the first 10 characters, then in the original string, there is a \r character that is dropped from the file.  The rest of the file doesn't match too well after that.  Also, the size of the strings are different too by a few thousand characters.  Is there some setting in the file read or write that could be causing this?  Thanks.
    Solved!
    Go to Solution.

    Here's a quick example (LabVIEW 2010). Result is always true in my limited testing.
    Here's the string IO version:
    (It is a bit more complicated when using the binary file IO:
    (1) wire false to prepend array or string size when writing.
    (2) wire -1 to count when reading.
    This code is not attached, just a picture)
    LabVIEW Champion . Do more with less code and in less time .
    Attachments:
    ValidateFileIO.png ‏5 KB
    ValidateFileIO.vi ‏7 KB
    ValidateFileIOBinary.png ‏5 KB

  • In Max, click Test Panel - Get Executable version (7.1.1) doesn't match resource file (7.1)

    Hi,
    I haven't used LabVIEW for a while.
    I plugged in a USB-6251 and ran Measurment and Automation explorer and saw that the USB-6251 was in the NI-DAQmx Device and was green.
    I clicked on the Test Panels button and got a window with the message:
    Get Executable version (7.1.1) doesn't match resource file (7.1)
    When I clicked OK  MAX was locked up.
    I found this:
    http://digital.ni.com/public.nsf/websearch/680E61A4D02158A186256F7A0073C228?OpenDocument
    which told me to repair the LabVIEW Runtime thingy.  I did that and it's still broken.
    My MAX version is 4.2.0.3001
    I assume that's a pretty old version but I would have thought it should have been updated when I upgraded to LabVIEW 8.2.1.  But I don't see a MAX folder in the 8.2.1 install.

    I thought I had replied to this but somehow it didn't get there.  In answer to the question about running the device driver CD after installing LV 8.2 I don't know the answer. 
    Anyway I installed LV8.6.1 and now it goes into test panels OK so I should be able to progress.

  • Bouncing individual tracks creates 2 wav files for each..why?!!

    When I bounce individually selected tracks from Logic Express, it creates a double wav file in my folder for each track I bounce. It didn't use to do that and I haven't changed any settings... Please HELP :/

    By "double wav" do you mean that you are getting a separate audio file for left channel and another for right channel, something like this?
    Be sure to check in your bounce window that you have "interleaved" checked instead of "split."
    Depending on how you're exporting/bouncing your tracks this may not help, but you can always solo the desired track and bounce your song with "interleaved" selected to get a single stereo file of that individual track.

  • My incoming mail user .wav file doesn't play for incoming mails. It does work on local test.

    I have windows 7 Professional PC and use Mozilla Thunderbird 24.4.0 as my email account. I have a local .wav music file for incoming mail which tests OK, but when a new mail arrives it doesn't play. I have all the boxes ticked in the settings. Can anyone help please?

    An update on local .wav sound file for incoming TB mail messages.
    Driver Booster found outdated drivers and all have been updated.
    Sound device has been removed from the device manager and automatically re-installed.
    From advice given I have turned mail.imap.use_status_for_biff to "false".
    Still no sound from my local .wav file. Is my problem unique? Is 2.74 MB too large as a sound file?

  • Playback of WAV files, doesn't automatically go to next track?

    I have a number of DTS-encode WAV files in a playlist that play fine thru my Airport Express, but unfortunately, it seems that iTunes doesn't automatically go from the end of one track onto the next track.
    How do I make it play a WAV playlist continuously?

    Mathew Hennessy wrote:
    How do I make it play a WAV playlist continuously?
    Make sure that all the little boxes (in the column at the left) have a check-mark in them.

  • My "Creative Cloud Files" folder in my computer doesn't match my "Assets/files" folder in the adobe website. How do I sinc the 2?

    My "Creative Cloud Files" folder in my computer doesn't match my "Assets/files" folder in the adobe website. How do I sinc the 2?

    Hi,
    Can you explain in more detail whats issues you are seeing exactly?.
    Thanks
    Warner

Maybe you are looking for