FCP Sound Syncing Question

Hey Everyone,
I have an audio syncing question for you. Any help would be greatly appreciated!
I am cutting in FCP. I logged and captured my footage and with timecode breaks, etc. the tape was cut up, and I was left with eight files, let's call them Footage1-Footage8. I have completed my cut of the piece using the in-camera audio that was linked to the video on the DV tape that I imported. Our sound guy in the field used a DAT recorder to record the production audio and apparently captured the timecode to the audio files, so that it matches the video (i.e. the DAT recorder and our cameras - DVX-100s had synced timecode during the shoot). The DVD he gave me has 20 audio files as .wav, let's call them Audio1-Audio20. My question is, now that I have finished my cut, I would like to replace the current audio tracks with the production audio. How can I sync the audio with the video and replace it in what I've cut. Or perhaps I can't, in which case - how can I sync them up and THEN cut, but I'd really rather have to not do that, lol!
Thanks for your advice!

Sorry about the long winded post, though.
I guess my real question, summarized is - is there a way to sync separately recorded production audio (that has timecode) with the footage that I shot (that has the same timecode as the audio, but also has in camera audio that accompanies the video track - I would imagine I can just "cut" the extra audio tracks, however).
I'm at my bay trying to figure this out, and I've got nothing, and the people I work for don't want to spring for that bwf2xml program you were talking about What's the emote for pulling one's hair out?

Similar Messages

  • Loss Of Sound Sync - FC7

    Hi Gang
    Working with FC7 in Snow Leopard 10.6.8 and an older Mac Pro.
    Exported a 70 minute clip and noticed loss of Sound Sync, (about mid way thru in a QT File). The clip is a mixture of assorted clips.
    Divided the 70 minute clip into 3 separate exported clips which produced proper 'Synced Sound' in each clip.
    This problem used to occasionally happen in FC6  - Does anyone know a fix or work-around?
    Thank You
    Mike

    "You've been in these forums long enough to know that FCP doesn't work well with non-FCP media"
    Shane - I don't appreciate your constant negative replies - do me a big favor, and avoid responding to any more of my posts.
    Yes, I've been here long enough to know you've always had a condescending attitude ....
    Your approach is NOT helpful. Its rude and disrespectful ....

  • Film sound sync

    Hey all,
    I'm doing my first film and sound sync. I wanted to make sure I was doing things the right way, as I am handing off the project to another post facility once the sync is done. We are both using FCP, aand I've got a call into the editor to make sure I'm getting to him what he needs. However, I wanted to run it by the experts (and elitists ;.) here to make sure I was doing it correctly.
    The film has been transfered to digibeta, and I have wav files with embedded timecode for audio. I'm assuming the embedded timecode, is location generated timecode in the portable dat or flash recorder that was used on-set. So the timecode doesn't match the timecode on the digi-beta. I will just use slate timecode to match to the timecode in audio file.
    So, what I planned on doing was just digitizing the digi-beta, logging it by scene. Then importing in the wav files. Then I'll watch for the slates in the footage and sync them in the sequence with the audio. Simple as that.
    Is there another way that I should be doing this? Is there a way to export the clips, so that I get a combined audio and video clip, that retains the timecode information of the digital beta?
    Thanks for any insight.
    Chris

    well, we usually sync in telecine for expediency in order to get sync rushes (dailes) to set but in addition we always load the location audio into the system (usually as BWAV files) and sync ourselves. that way we keep control of all the audio media through the post process i.e. we copy the audio media onto a drive for the sound dept and when the time comes we give them reference OMFs and they can reconnect to the master audio tracks.
    this is mainly due to the way we work with PAL in the UK (we telecine fast at 25fps) and also the fact that we usually telecine to Beta SP rather than Digibeta for offline purposes.
    mind you the current film i'm working on here is NTSC (it began life in Canada) and we have all the original audio in the system.

  • Just installed FCP 10.1.2 and my FCP sound effects folder is missing

    I just downloaded and installed FCP X 10.1.2 and was noticing that under Music and Sound that my FCP sound effects folder is missing. Any suggestions as to how to get it on there. Thanks for any help.

    Yes i went to additional content, it said everything up to date. I also did the quit FCP and then start Garage Band and quit it and restarted FCP still no FCP sound effects folder

  • FCP to PP questions: part 3

    7) In FCP Shift+ALT allows me to drag a duplication of a clip onto another track, giving me two clips of the same content.
    Did this get added to PP6? If so, anyone know the keyboard shortcut. If not, feature request?
    8) When I open a bin and have them clips set to thumbnail, why are the clips in reverse order and how do i swap it to alphabetical order? In list mode they are in alphabetical order.
    Sorry this is a real dumb question but i can't seem to see what to click to swap the order of the clips in thumbnails.
    9) In FCP I have f5 set to lock all audio and f6 set to lock all video. This means if i only want the audio or the video off a clip it is easy to click the other key shortcut and close off what I don't want.
    I can't seem to find a lock all audio and lock all video shortcut. Are they there and if so where?

    8) Icon mode, as Jim said, is a manual sort mode - the idea being it can be used for storyboarding, so you can sort the clips in rough presentation order, select them & automate to timeline (or drag to the new sequence button to create a new one & it'll auto fill the new sequence with the clips in the order you selected).  Having an auto sort in Icon mode is a feature request that I've seen several times in the past few week, so we're definitely aware, but it's always good to add your voice via our feature request page, as we pay attention to the relative popularity of the requests that come through.
    9) I wasn't quite following the context here.  Jim's talking about if a linked A/V clip is already on the timeline, if you alt-click the video or the audio portion, only that portion will be selected.  Then (letting go of alt), you can drag just that selected video or audio segment if you need to deliberately slip them out of sync (and you'll get out of sync indicators showing up to tell you how much of a slip you're introducing).
    There's other locks as well - you can lock individual tracks, so if you lock an audio track, you can select a group of linked video clips and have everything slip relative to the locked audio track if that's what you're looking for.
    If you're talking about inserting from the source window into a sequence, then you have a few options as well:  it sounds like you've already noticed the video only/audio only icons for dragging, that's one way;  you can also use track targetting in combination with the insert/overlay shortcuts, so I could for instance disable all audio tracks in the track targetting, and my insert would then only bring over the video.  I'm probably forgetting other methods - multiple ways to skin cats here.
    Cheers

  • Subframe Audio Sync Question

    I'm syncing audio recorded from a separate source, and the shot is slated.
    I want to sync the waveform to picture as close to 1/100th as possible.
    I didn't think I would ever need to ask a question like this, but there's a debate brewing here, so here it goes. The question is more about how FCP works and how it plays the single frame, where when zoomed all the way in and you can see the dark area representing that single frame. The playhead shows the beginning of the frame, but doesn't it play through the audio waveform of the single frame while that frame is "paused" for 1/24th of a second?
    There are two theories we're seeing out there; one is to sync the waveform at the frame where you see the slate is closed, even though it happened in the space between the previous frame. The other is that you can reasonably estimate where the slate would have hit between frames and allow the waveform to land just before the frame where the slate is closed.
    For example, say you have a frame where the slate is just about to hit and it was moving fast, so fast there's motion blur, but there's a small enough space where you know the sound hasn't happened yet, maybe only a centimeter or less. Obviously the next frame shows the slate closed, and you can guess the slate actually touched between the time each frame was exposed, probably even closer to the previous frame.
    So, where exactly to place the waveform? At the beginning of the frame where it's closed, or at the end of the previous frame where it would have happened in real life?
    Here's a silly attempt at an illustration of the dark area representing a single frame on the timeline and the waveform underneath:
    {Single frame}{Single frame}
    .................||......................
    {Single frame}{Single frame}
    .....................||..................
    (The || represents the waveform generated by the slate.)
    If anyone could weigh in on this and settle the debate we would appreciate it! Thanks in advance.

    I've dealt with this issue in some detail when assisting and I'd say your thinking is sound (no pun intended) regarding the way fcp displays a single frame.
    We were moving to a deva recorder using synced timecode and were getting consistently confusing results where the audio seemed to be in advance of the picture. When we tested by recording directly to the camera we found that the behaviour we were seeing was normal ie the audible clap often occurred on the frame before the sticks were visibly closed. Obviously on previous series, where we'd used the sound recordist's stereo mix track from the camera tape, we'd never had any reason to pay attention to the sync, we weren't doing any syncing so we just assumed it was right. When I looked back at those old rushes tapes I found that this behaviour had always been the case.
    d-trick wrote:
    For example, say you have a frame where the slate is just about to hit and it was moving fast, so fast there's motion blur, but there's a small enough space where you know the sound hasn't happened yet, maybe only a centimeter or less. Obviously the next frame shows the slate closed, and you can guess the slate actually touched between the time each frame was exposed, probably even closer to the previous frame.
    So, where exactly to place the waveform? At the beginning of the frame where it's closed, or at the end of the previous frame where it would have happened in real life?
    Here's a silly attempt at an illustration of the dark area representing a single frame on the timeline and the waveform underneath:
    {Single frame}{Single frame}
    .................||......................
    {Single frame}{Single frame}
    .....................||..................
    So in this case I'd say the first is 'correct' for film or interlaced video at a high shutter speed. At low shutter speed I'd say it's a toss up, it's too hard to tell if the sticks shut in the time between frame 1 and frame 2 being exposed or just too early in frame 2 to produce any detectable motion blur. That said, I don't think anyone would be able to tell the difference between the two anyway.
    Of course in an app that doesn't support subframe editing the alternatives would actually be
    {Single frame}{Single frame}
    .................||......................
    {Single frame}{Single frame}
    ......................................||.
    So the second option is clearly delayed which makes it a no-brainer, and saves you the agony of choice.

  • FCP to HDCAM Question!

    Hello,
    I'm cutting a 4 min short film in FCP 6.0.5. We've recently been accepted to a fest that wants HDCAM. We shot on the XDCamEX1 1080p24 (35mb/s) and set our sequence settings to match this.
    I have two questions:
    1. I'm doing the mix myself in FCP, with some sweetening in SoundtrackPro. It is a fairly simple mix, but how should I monitor the overall sound? The loudest point in the film hits about -6db on FCP's audio meters. Is that correct? Also, we're projecting in a large theater, so I've been mixing with high quality headphones in order to hear any hiccups within our production audio. It sounds decent. Do you think we will be okay or theater speakers projecting a stereo mix pick up more discrepancies?
    2. I'm color correcting the film myself. Considering our master will be on HDCAM, should I use my 720p LCD TV as an external monitor or my 23" 1920x1080 apple cinema display? Those are the only two options I have. What pitfalls should I look out for?
    Thanks for any help you can give.

    Well, if it were me and these were my options, I would probably use the cinema display to color correct because at least I'd be able to get a pixel for pixel simulation of the picture since you're editing 1080p.
    However, when you output, I'd rent an HD QC monitor in addition to the HDCam deck to double check my work and ensure a good-looking master. You won't be able to control the environment that it's being projected in, so your best bet is to produce a good-looking tape.
    Andy
    PS: As for the audio, find out if the festival has tech specs on audio levels. Also, it's ok to mix with headphones on, but you'll really need to hear the film on speakers. Preferably a good set, and an old, beat up set. That will give you perspective on how it's likely to sound in the theater.

  • FCP 5 & Motion2 question

    Hi All, hope this question isn't a silly one.
    When I go to the send to (motion) option in FCP the media goes off-line, once I am in Motion I get asked the question to reconnect, however the name of the file it asks for is not the exact same one I am working on, for example if I am working on Jack & Jills video it will ask me to reconnect to Jack. sorry for sounding silly but it is the best way for me to explain it. Any help would be great.

    Hi Patrick
    I will try dropping any non alphanumeric characters today, I feel confident that this could be the answer to my problem.
    Thanks Alot, and I will get back to you with the result.

  • Imovie 9.0.2 sound sync - is it to do with the export route

    Like scores of others in these forums I've been grappling with the (appalling) failure of iMovie 9 to synchronise sound and vision. The 9.0.2 update did not solve anything for me.
    I have founds one solution, though not sure how good a one. Instead of using "export using Quicktime" - which produces an out-of-sync mov file - I find that "export movie" gives me an m4v file that does appear to be in sync.
    I've no idea what light this might throw on the whole sorry saga. 2 questions for me at this stage:
    1 - am I losing quality or creating other problems by saving as an m4v?
    2 - how do I go about compressing my m4v (it's 30 times the size of the equivalent mov file)?

    DV
    4 minute and 34 minute projects - length seems to make no difference. Also they are out of sync from the outset - not just by the end as some users have found.

  • Sound Syncing Strangely

    Hello friends.
    I am currently making a film using Flash 8 Pro. In one of my
    scenes, I placed and synced the sound exactly as I did in all the
    others. I set the sound to Event, because I'm using a Virtual
    Camera. When I exported the video, the sound was not properly
    synced (all sounds happened much sooner than they were supposed
    to). I tried moving the sound's keyframes so they could occur later
    (I thought that this problem was a result of visual lag), but the
    sound occurred at the exact same time it did before I moved it. I
    tried restarting Flash, saving and re-exporting, deleting the sound
    files and re-importing them, moving them to a different layer, and
    copying everything to a brand new file, but nothing has worked.
    This problem is only occurring in one scene file, very
    strange.
    Thanks.
    - Wright

    Make sure your default easy setup frame rate matches your camera material.  That can supposedly cause this problem.  Open one of the audio files in fcp and check the frame rate and see if it matches the video.

  • Sound Sync issues

    Hi there. Can anyone help me with this. I've recently captured footage from DVCPRO HD tapes (1080i25p) to the DVCPRO HD format (1080 25i) in FCP through my KONA 3. The footage i have of an interview seems to be out of synch with the sound...not sure by how much, but after doing what i'll explain next, i'm betting it's about 4 frames!
    I wasn't too concerned as it's something i've noticed before and thought that i could have made a mistake with the capture (the KONA 3 is new) and i'm fairly happy to re-sync the sound if needs be. The main problem is that i have now opened an old project (DV PAL) that needs a number of sequences printed to DV tape for my client. Upon opening my project it seems that all the sync to the interviews in the project are now 4 frames out!! Very annoying!
    I can't find anything that could be wrong here within my settings and can't quite get to grips with it but suffice to say, i could do with a pretty quick solution!
    I don't have my RAID turned on with my other project and am just running this from a firewire 800 drive. The project has not been touched since about a week ago when everything was fine!! I've tried deleting all my render files for the project but this hasn't helped either.
    If anyone can help with this i'd really appreciate it
    thanks
    Alex

    Thanks Ken, I've managed to resolve the issue to a certain extent now. There did seem to be some sort of lag introduced by the Kona even though i wasn't directing sound through it or even had anything plugged in to the sound out connectors on the breakout box. I disconnected the HDCAM deck i have in the suite from the breakout box and things are now back in sync. A solution, but certainly would be interested to know why as I'd like to think i'd be able to cut stuff even if a deck is connected to my breakout box.

  • Sound sync problem - unsolvable?!

    Hi,
    Quite a basic one I suppose: I'm editing a video source with a separately recorded sound source of the same event. The sound of the footage (HDV unfortunately) is 48kHz, while the separate source is 44.1kHz, so obviously if I sync them up at the beginning, before long they're out of sync again.
    Is there any way around this? Can I transcode (if that's the right word) the 44.1 up to 48? Or should I work the other way around, bringing the 48 down to 44.1?
    Is there a solution to this problem? It kind of buggers the entire project if I can't sort this out!
    Looking forward to your help,
    Steve

    Common solution is to convert the 44k to 48, using iTunes, or you should also be able to do it in FCP...might depend on what format the 44kHz file is.
    From FCP, you could do an 'Export as Aiff/16bit/48kHz', then import that file for use in the project.
    I don't use the iTunes method enough to detail the process, but I remember that if I was using say...CD Audio (44kHz), I'd simply set iTunes preferences to automatically convert CD Audio to AIFF/48kHz...something like that.
    Hope that helps,
    K
    K

  • K7N2 Delta Onboard Sound Disable Question

    I have a K7N2 Delta motherboard with a MSI FX5600 256MB DDR video card.  I am currently using the onboard audio.  Ever since I first built this computer I have been having nearly every game I play on it lock up and freeze.  I am thinking that it is a sound issue that is causing this problem, so I want to install a Creative Labs Sound Blaster Live.  I have several questions that I hope someone can answer as I have exhausted all avenues and still don't have any answers.  Here are my questions:
       1.  How do I disable the onboard sound?  ie  Which jumper is it and what are the
            jumper settings?
       2.  Which PCI slot is recommended for installing the sound card so that I won't
            any IRQ conflicts?
       3.  Is this actually an audio problem or is there something else wrong with this
            motherboard or video card?
    This is becoming very frustrating because when the computer locks up all I can do is just reset without shutting down.  I'm becoming more and more worried that if this continues unabated I'm going to crash the hard drive and have to replace it only to have the problem persist.  I hope someone can give me specific answers to my questions.  Thank you in advance for your help.

    First. Is it K7N2 Delta-L or Delta-ILSR (the one with an actual Soundstorm chip on it)? If you have the actual Soundstorm chip, use it! It will sound much better than an SBLive. Assuming you have the normal K7N2 Delta-L like I have:
    1. To disable the onboard sound, first uninstall the drivers for it, then you need to go into the BIOS by holding the Delete key while you boot up. Then find the section "Integrated Peripherals" and in that section "Onboard Devices". Press Enter and find AC97 Audio and disable it. There are no jumpers.
    2. PCI slots 1, 2 and 3 are usually not shared with other stuff like you AGP or USB ports. PCI Slot 3 is the ideal choice. 1 is usually covered by your video card fan anyway. For the SBLive, on Windows XP/2000, get the newest drivers from Creative Labs. The file is named "LiveDrvUni-Pack(ENG).exe" and it's file size is 23.3 MB (24,451,404 bytes). They have fixed the "nforce2 + gameport incompatibility".
    3. The problem may be audio related...or not. Which drivers are/were you using for your onboard audio? nvidia or Realtek? Actual Soundstorm chip or crappy software codec? It could also be video card related. Fastwrites? 8x or 4x AGP?

  • Sound looping question

    hello everyone!
    i'm pretty new to using flash (i only need to use it for a uni sound design project) and i'm having some problems.
    what i'd like to do is create a button, and loop sounds in the "up" state of that button. the only problem is, that if i import the sound to the "up" keyframe and choose loop in the properties it keeps looping until the end of time. what i'd like though is it looping just in the event of the mouse being in the buttons' area. what should i do?

    1) create button
    2) doubleclick on the button (edit)
    3) create extra layer for sounds
    4) in this layer, create empty keyframes for all states (UP, OVER, DOWN, HIT);
    5) add the sound to the OVER keyframe in sounds layer, set it to SYNC: Start, Loop
    6) add the sound to the UP keyframe in sounds layer, set it to SYNC: Stop, Loop

  • Log file sync question

    Metalink note 34592.1 has been mentioned several times in this forum as well as elsewhere, notably here
    http://christianbilien.wordpress.com/2008/02/12/the-%E2%80%9Clog-file-sync%E2%80%9D-wait-event-is-not-always-spent-waiting-for-an-io/
    The question I have relates to the stated breakdown of 'log file sync' wait event:
    1. Wakeup LGWR if idle
    2. LGWR gathers the redo to be written and issue the I/O
    3. Time for the log write I/O to complete
    4. LGWR I/O post processing
    5. LGWR posting the foreground/user session that the write has completed
    6. Foreground/user session wakeup
    Since the note says that the system 'read write' statistic includes steps 2 and 3, the suggestion is that the difference between it and 'log file sync' is due to CPU related work on steps 1, 4, 5 and 6 (or on waiting on the CPU run queue).
    Christian's article, quoted above, theorises about 'CPU storms' and the Metalink note also suggests that steps 5 and 6 could be costly.
    However, my understanding of how LGWR works is that if it is already in the process of writing out one set of blocks (let us say associated with a commit of transaction 'X' amongst others) at the time a another transaction (call it transaction 'Y') commits, then LGWR will not commence the write of the commit for transaction 'Y' until the I/Os associated with the commit of transaction 'X' complete.
    So, if I have an average 'redo write' time of, say, 12ms and a 'log file sync' time of, say 34ms (yes, of course these are real numbers :-)) then I would have thought that this 22ms delay was due at least partly to LGWR 'falling behind' in it's work.
    Nonetheless, it seems to me that this extra delay could only be a maximum of 12ms so this still leaves 10ms (34 - 12 -12) that can only be accounted for by CPU usage.
    Clearly, my analsys contains a lot of conjecture, hence this note.
    Can anybody point me in the direction of some facts?

    Tony Hasler wrote:
    Metalink note 34592.1 has been mentioned several times in this forum as well as elsewhere, notably here
    http://christianbilien.wordpress.com/2008/02/12/the-%E2%80%9Clog-file-sync%E2%80%9D-wait-event-is-not-always-spent-waiting-for-an-io/
    The question I have relates to the stated breakdown of 'log file sync' wait event:
    1. Wakeup LGWR if idle
    2. LGWR gathers the redo to be written and issue the I/O
    3. Time for the log write I/O to complete
    4. LGWR I/O post processing
    5. LGWR posting the foreground/user session that the write has completed
    6. Foreground/user session wakeup
    Since the note says that the system 'read write' statistic includes steps 2 and 3, the suggestion is that the difference between it and 'log file sync' is due to CPU related work on steps 1, 4, 5 and 6 (or on waiting on the CPU run queue).
    Christian's article, quoted above, theorises about 'CPU storms' and the Metalink note also suggests that steps 5 and 6 could be costly.
    However, my understanding of how LGWR works is that if it is already in the process of writing out one set of blocks (let us say associated with a commit of transaction 'X' amongst others) at the time a another transaction (call it transaction 'Y') commits, then LGWR will not commence the write of the commit for transaction 'Y' until the I/Os associated with the commit of transaction 'X' complete.
    So, if I have an average 'redo write' time of, say, 12ms and a 'log file sync' time of, say 34ms (yes, of course these are real numbers :-)) then I would have thought that this 22ms delay was due at least partly to LGWR 'falling behind' in it's work.
    Nonetheless, it seems to me that this extra delay could only be a maximum of 12ms so this still leaves 10ms (34 - 12 -12) that can only be accounted for by CPU usage.
    Clearly, my analsys contains a lot of conjecture, hence this note.
    Can anybody point me in the direction of some facts?It depends on what you mean by facts - presumably only the people who wrote the code know what really happens, the rest of us have to guess.
    You're right about point 1 in the MOS note: it should include "or wait for current lgwr write and posts to complete".
    This means, of course, that your session could see its "log file sync" taking twice the "redo write time" because it posted lgwr just after lgwr has started to write - so you have to wait two write and post cycles. Generally the statistical effects will reduce this extreme case.
    You've been pointed to the two best bits of advice on the internet: As Kevin points out, if you have lgwr posting a lot of processes in one go it may stall as they wake up, so the batch of waiting processes has to wait extra time; and as Riyaj points out - there's always dtrace (et al.) if you want to see what's really happening. (Tanel has some similar notes, I think, on LFS).
    If you're stuck with Oracle diagnostics only then:
    redo size / redo synch writes for sessions will tell you the typical "commit size"
    redo size + redo wastage / redo writes for lgwr will tell you the typical redo write size
    If you have a significant number of small processes "commit sizes" per write (more than CPU count, say) then you may be looking at Kevin's storm.
    Watch out for a small number of sessions with large commit sizes running in parallel with a large number of sessions with small commit sizes - this could make all the "small" processes run at the speed of the "large" processes.
    It's always worth looking at the event histogram for the critical wait events to see if their patterns offer any insights.
    Regards
    Jonathan Lewis

Maybe you are looking for

  • Is it possible to get document NO. create by bapi BAPI_COPAACTUALS_POSTCOST

    Hello experts,   in my ABAP program design, i want to trans data through the bapi:BAPI_COPAACTUALS_POSTCOSTDATA to trans data from a SCV file to the SAP CO-PA, the user want to see the  error infor if the BAPI does'nt success, if the BAPI is success,

  • Intermittent Wireless Connection, Dropping Wireless Connection

    I am asking this in hopes of finding at least some way to diagnose my problem and hopefully eventually solving it. My wireless connection currently drops at varying amounts of time. Sometimes I can go 20mins others merely 5mins. The Airport reads tha

  • Swap primary secondary role in Cisco ASA 7.2

    I have2  pairs of cisco 5520 and cisco 5550 running 7.2 software working in active/standby mode in single context mode. I want to  change the primary and secondary roles of firewalls without any downtime. What is the best way to do this? failover exe

  • How can I report my stolen iphone

    I want to report my stolen iphone!!!!! It is not to easy to report!!! how can I report it??

  • For Update and Where Current Of -

    HI All, We have been using Select For Update statement with wait time (i.e in seconds) in various places inside our applications. This forces for locking those rows and we commit the work at the end of our work to ensure we release the locked rows to