Adjust multicam audio phase/timing

Hi All
I do lots of multicam concerts, and don't have genlock or timeslave on my cameras. Therefore the audio in my multicam sequences is often out of sync by a fraction of a frame. About half the time, this creates a noticeable "phase" effect when I mix audio from two or more cameras. Is there a way I can nudge the timing of the audio on one track by a fraction of a frame, in order to eliminate the effect? A similar phenomenon occurs on audio from a single camera, when the audio is coming through different sources (e.g. A1 from a wireless on stage, passing through a mixer, alongside A2 from a shotgun, wired straight into the camera). I'd really like to pull this off without a round-trip to Audition. (I am using
PrPro CS6 on Win7)
Thanks!
Eric D - CoralVision

If you check Show Audio Time Units in the sequence flyout menu, you will be able to nudge audio clips at the sample level.

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An investigation of times required to burn DVD during 'audio' phase.

These figures are for a 1 hour 40 minutes movie, with 3 tracks of Audio, trying to save to disk image on desktop. 80 Gig of HD free.
When iDVD appears ‘hung’ it is actually saving to a file, called ‘tempmovie.mov’, in the iDVD project, which in turn is actually a package. I used a spreadsheet to track the size over time of this file, which hit a max of 1.01 GB in my case.
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Rendering Menu/Assets took 3 1/2 hours.
Rendering Audio
- Phase 1. iDVD ‘locked’ and ‘unresponsive’. Rendering the file tempmovie.mov took 42 hours using up to 80% of processor time. iMovie used 12-18%. Initial spurt of several hundred Mb of file size in an hour, then about 100 Mb in second hour, then adding to the file at a low of 10 Mb per hour, high of 15 Mb/hr.
- Phase 2. Spinning activity wheel (burning wheel?) started back up after indeterminate time (running at night). New file called ‘CurrentAudio’ created, but not added to, no files ‘growing’. iDVD unresponsive, but not marked in the Activity Monitor as such. Menus greyed out.
Rest of audio terminated after waiting 60 hrs with iDVD using average of 13% processor time. I gave up on it, had to force-quit. iMovie was using about 8% of the processor time.
• Second attempt...Single audio track.
Trashed old iDVD project. Opened iMovie file with Garageband; took 41 hours processing time to merge the 1 stereo track and two mono tracks to 1 stereo track. Saved audio as two stereo tracks of 192 bit AIFF in less than 10 minutes. Exported it to iTunes.
Turned off two tracks in iMovie file (top & bottom tracks), erased third (center) track and imported Garageband track to center via iTunes. Sent to iDVD.
Rendering Menu/Assets took 3 1/2 hours
Rendering Audio
- Phase 1. iDVD ‘locked’ and ‘unresponsive’. Rendering tempmovie.mov took 16 1/2 hours, max. 80% processor for iDVD, max.12% for iMovie. Initial spurt of 240 Mb file size in first hour, then about 130 Mb in second, 90 in third hour, 66 in fourth, average of 45 +/- 6 Mb per hour after 4 hours. At end, 20Mb file ‘CurrentAudio’ created.
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Trashed old iDVD project. Made small edit to iMovie project. Changed destination for finished disk image to Movie folder rather than desktop.
Rendering Menu/Assets took 3 1/2 hours
Rendering Audio
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- Phase 2. 11 minutes to save disk image.
Conclusions.
Using three tracks of Audio increases the rendering time enormously (42 hours for audio).
It also pays to erase unwanted tracks in iMovie rather than just unticking them (16 hrs vers. 1.75 hrs for audio).
It takes the same amount of time to render three tracks in iDVD as it does to export them to Garageband, (and you still have to re-render in iDVD afterward).
Either iDVD doesn’t like saving to the desktop, or I had a corrupt/missing file that prevented saving the disk image. Probably the latter.
Regards
Santa
2003 sp 1.8 GHz G5 1.5 Gb Mac OS X (10.4.8) 3 x 300Gb Seagates, Epson 3490 scanner, Epson 830 printer

Brian,
this year's price "Most Longanimous Person of The Year" belongs to you!! THANKS for you efforts and reports..
you've probably read Karl Petersen's reports about some mysterious "benchmark", when an iM project gets stuttering/unrespondsively due to "too many" audio clips..
.. I'm actually producing my Xmas gifts.. 55min project = 2h encoding ... (see spec below) ...
but: I'm editing my iM projects mostly on an external harddrive; and it is a "habitual ritual" of mine, to import the final movie as QT/Full Quality file onto my internal hd, THEN fire-up iDVD => never had any trouble with audio-encoding; my videos are "solid" dv-streams ...
so, how about adding that (single dv-stream of your movie) setting to your test set-up ...?
and: why the hack 1:40h for testing?? why not ... 5min? ;-))
happy holidays ..

System Copy -Solman 7.1 Adjust name table phase error

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I am on way of performing the system copy for solman 7.1 from prod to dev system.We are encountering the error in the adjust name table phase .Below is the log :
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SysLog:hBZA20140515082322000000000000 this TTY
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Hi Asim,
1.I was getting error at Abap phase load error .So i have merge the tsk & bak files by r3laod and manually set ign in tsk files .Should i change the ign status back to error.Is this could be the problem causing in adjusting table name phase.
Status "ign" will ignore the object from being imported into the database. This may lead to inconsistency and loss of data.
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Regards,
Deepak Kori

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.
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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.
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SoundTest.java
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import java.io.*;
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SoundElement.java
import java.io.*;
import javax.sound.sampled.*;
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        soundTest = soundTestIn;
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        clickFrequency = clickFrequencyIn;
        subdivision = subdivisionIn;
        generateAudioFile();
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        audioFile = new byte[clickLength * 2];
        double temp;
        short maxSample;
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            temp = Math.sin(2 * Math.PI * p++ / (soundTest.sampleRate/clickFrequency));
            maxSample = (short) (temp * Short.MAX_VALUE);
            audioFile[i++] = (byte) (maxSample & 0x00ff);
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        createPlayer();
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        while (soundTest.playing){
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Thanks but you have never posted reply s0lutions before ?? And F 4 is definitely not 10 times faster as stated before I upgraded !!

Adjusting different audio clips levels?

Working in FCP and have a project with a lot of small individual clips of people speaking and each clip is at a different audio level. What's the best and easiest way to adjust the audio on each individual clip without over doing it labor wise?
Thanks
RD

Open each clip in Soundtrack Pro and normalize them to -6db. That will give you a reliable baseline for all the clips, and put the peak levels right where you want them.
Then, either in STP or back in FCP, adjust the level of each clip so the average dialog level is around -12db. Do this both by ear and watching the audio meter.
I suppose in FCP you could do the avg level adj. by keyframing the audio level, too. That would stay with the clip before you put it in a timeline. To use the faders you need to have the clip already placed in the sequence timeline.
If you find you need to raise the level of the normalized clip to get the avg. level to -12db, then you better put a limiter on the clip to keep the peak below -6db.

When adjusting the audio volume in a selected region, why does it give me a curved line instead of straight and why cant I change it like i can with a fade in our out?

When adjusting the audio volume in a selected region, why does it give me a curved line as a transition instead of straight, and why cant I change the fade shape like i can with a fade in our out? When I slide thet volume all the way down it makes a very abrupt down curve. It's audibly noticdible.
Thank you.

Use the range selection tool and select the portion of the clip you want to change the volume on. Drag the volume line down and FCP will create keyframes which you can adjust as you want to get a fade shape.

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?
>

What do you do if you need to adjust the audio mid clip?

In Final cut X can you adjust the audio in the middle of a clip? It seems that the fades are only at the beginning and end of each clip. What do you do if you need to adjust the audio mid clip?

Range tool (click R)
select range
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Forum,
I'm having to adjust my audio levels in the viewer or timeline one channel at a time. How can I set FCP up so that both channels adjust in unison? Twice as fast you know!
Trey

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I have my iMac hooked up to my TV and when I watch movies from it the audio levels very quite a bit. Some parts are really loud and some are very quiet. Is there a program that can auto adjust the audio levels out of the mac?
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no help

Multicam audio levels/meters??? - Premiere CS6

Is there a way to see the audio meters while in the multicam window?
The audio plays fine but is there really no way to monitor the audio to see if its peaking or not, or what level its at, unless I stop the multicam and play it in the timeline?
All the source clips, all the timeline clips, all the project window clips all enable the audio levels/meters.
Why not the multicam? Seems odd.

Audio is the other half of what editors deal with so I kinda do need to worry about it.
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Jim Simon wrote:
I'm starting to learn that you are the king of "that's not what I care about so it must not be important".
It's actually more often a "how can I get this job done with the tools and features that are available to me right now" kind of attitude.
While that's a practical approach and certainly one that will cut down on stress, that's also the mindset that allowed for companies like RIM (Blackberry) to fall behind their competitors. I prefer a more "work with what you've got but if something's missing or broken, speak up" approach. My hope is that the PPro design team actually listens to their users to make it better. We all have things we wish would improve or be fixed and if we all just said "eh, it is what it is" then nothing would ever improve. I'm more of a "if it ain't broke, don't fix it...but if it is, then stop making excuses and fix it". I know the absence of something as simple as audio meters doesn't mean mulitcam is broken, but its an example of an oversight on the part of Premiere's design team that I hope they fix. It's a very simple feature. Every other NLE I've worked with has it. It helps the workflow especially with long multicam sequences. There's no reason it shouldn't work like it does if you play a clip anywhere else. Now its just a matter of if they care enough to fix it.
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Multicam audio

I'm a long time user of FCP 7 but I'm new to FCP X. I've now spent two days searching tutorials and help pages for an answer to no avail.
I'm running a small studio where we're doing many, many on-set three camera interviews (Cam 1, Cam 2 and Cam 3). All the audio is on the interviewee's camera - Cam 3.
I'm good with creating multicam clips, I'm good with designating all the audo from Cam 3 and I'm good with selecting "Video Only" so I'm just cutting the video between Cams 1, 2 and 3 while keeping the audio from Cam 3 throughout.
My problem is that although the Cam 3 audio is continuous, there's a break in the track every time I switch video between Cams 1, 2 and 3. That makes it time consuming to adjust audio levels.
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TIA,
Peter

My problem is that I'm checkerboarding a discussion; lowering the level on the person who isn't speaking when they're silent. Changes to volume aren't likely to be global but to follow the back and forth of the discussion. Someone who's excited might interject at a high volume then lower that volume when they have the floor. That adjustment in volume often spans a cut from one camera to another. Ideally, the audio track would be continuous and not chopped up into little pieces. I'm beginning to think that the only solution is to detach the audio.

Multicam Audio Sync Open in Timeline

Hi everyone. I will try to keep this simple.
I have a 2 camera setup, each with audio plus one external audio track. (3 total audio tracks)
When I create a multicam source sequence I sync via audio. I then open the source sequence in the timeline to make sync adjustments. However, it only plays back audio from Track 1. When I place this source sequence on a multicam Target sequence, I get all of the audio tracks to play.
My question is "Can I not get all of the audio tracks to play when I click "Open in Timeline"?"
I've been reading on this all day and can't seem to get anywhere.
Thanks for your help.

If you want all audio at once, after you multi cam audio, you'll have to alt\opt+leftclick+drag the audio down below the master to duplicate it (do this once for every Camera). You then do the alt\opt+rightclick, go to multi cam and select a different camera for each of the tracks. Now all audio will play.
NOTE: With one track you can cut through with razor, and switch cams where you cut. If you cut your multi cam the automatic style, cuts will be made in audio track as well, that line up with video switch. You can adjust these edit marks to your desire the same way you would when using trim or roll, and you could switch Cameras from the alt\opt+rightclick menu. Using automatic mode to cut video, you will cut all LINKED audio at the same point. You can apply this to multiple tracks by linking them all to the video track before you start your camswitching. Afterward, you could adjust, delete, trim, roll and switch through your audio to your hearts content.

Multicam audio disappeared!

So I had a short multicam project put together. 2 cameras with audio from a Tascam audio recorder. Set the multicam file to read the clip from the Tascam as the main audio stream. All good. That is until I double clicked the multicam file, added some gain, and adjusted the settings of the audio file. I thought this would simply modify all the cuts of the multicam file in my main project, but instead FCPX removed all the audio in the main project, waveforms and all. I can re-assign a different audio stream, and that shows up, but if I assign the Tascam file, all audio is gone. Ive tried reverting away from the changes I made to the audio settings and effects to no avail.
Any ideas how to reverse this?

Found the issue. Some how when I was modifying the audio track within the multicam clip, the audio itself got moved to the movie track directly above it. So, instead of audio being in the track labeled Tascam 032, it was on the track labeled MVI_3670. Don't know how it happened, because I couldn't move the audio back down to Tascam 032.
Either way. Found out. Fixed. Done. If it happens to someone else, take a look at the tracks.

Adjust multicam audio phase/timing

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