Question about audio channels in FCP

Because of the way audio was captured with a few clips in my sequence, the clips audio is only coming through one channel (speaker). The other channel contains nothing. How can I duplicate the channel with audio and assign it to the other speaker?

I did notice that. There are a small few of us on here.
So now that I've helped you out, bring me some Great Plains Sauce & Dough.

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4 audio channel from FCP to miniDV

Hi There
I have to print a tape from FCP with 4 audio channel but don't know how to do it. Does anyone know? Thank you.

Ah...there you go then.
I haven't done it myself. Maybe someone else can offer nuggets of info...
Shane

About audio channels, need help... :(

elo people, im trying to make two audio channels in a video, how can you do it, where if it where a normal video, you will be able to just choose the left side or the right side of the audio...like in a karaoke, where youcan take the singing voice off yet still hear the music playing...how do i do that? i have a minus one file and a norml file in which a singer sings...how do i incorporate that in the video??? tnx, and godbless

i was able to make both the left channel and the right channel, but when i exported the video, i couldn't find the left and right channels..how is that? do i have to do something before i export the video?

This is a very simple question about multi-channel audio playback

I have an mp4 file that i made and i made it 7.1 surround sound, and i'm pretty sure that this 7.1 surround sound works, as it can be played in VLC. i'm using netstream to load my files now. How do i make it so that flash can playback all 8-channels of sound? I suppose kglad would know the answer

Flash doesn't support multi-channel audio (yet?)

Question about audio in FCP

When I import an audiofile that is mastered to -12dbFS the meter in FCP shows about -18db..
Why?

how was this file mastered?
when i record a clip with -20db tone, and then import that into fcp it registers at -20db, exactly as it should

4 simple questions(I guess) about audio files in FCP X.

Hello,please help me.
1) Is it appropriate to import my audio files from "Music and Sound Browser" tab into FCPX) or its a better idea to arrange all Audio files
what I am going to use in a folder and then import from finder window?
2)Is it possible to drag and drop music tracks from "Music and Sound Browser " tab to Event Browser without put the audio
in sequence?how?
3)How to import a part of a music / soundtrack? (Can I mark in / out BEFORE I drag to timeline)? I mean,set IN / OUT directly from
Music and Sound Browser" tab.
4)How to connect music / sounds what I have in "Music and Sound Browser" tab TO connected clips (NOT to primary story line clips)?
thank you

Actually, you can have music from the Media Browser edit any way you want into a Timeline: either to a Storyline, or be a Connected Clip, or do an Insert Edit. Highlight the song, click the desired Edit button. Keyboard shortcuts don't work for this.
Drag-and-drop any selection for the Media Browser to an Event. It will COPY that audio file into your Event folder, so you'll now have two physical copies. One in iTunes/sound library, one in your Event folder.

Question about BlazeDS Channel internals

There is much useful information available about how an application developer would use BlazeDS to implement their application, but there is a shortage of information about how BlazeDS is implemented. There are very general architecture diagrams, but no information on the actual network technologies employed. This information is essential for forming estimates of the server hardware requirements of a design. The alternative is to build the application one way, and then discover the performance characteristics afterwards.
For one feature of my application, the ideal structure would be to give the clients and mx.Consumer to receive non-polling stream of changes replicated from the database. In this particular feature, there would be no subtopics, each client would receive the same data. The question is would BlazeDS/Flex give each client 1 persistent socket to support the stream. If this is the case, then this design poses a very significant load on the server. It makes it easy for the developer, because he writes to only one topic, but if under the covers, the server is writing to one socket for each client. It will require a lot of hardware.
Or, to state the question another way, would BlazeDS/flex use multicast, when there is a client Consumer subscribing to a topic, with no subtopics?

I have just found one clear statement in the BlazeDS documentation with regard to sockets when streaming:
"The Servlet API uses blocking IO so as with long polling above, you must set a configured upper bound on the number of streaming connections you allow. Clients that exceed this limit are not able to open a streaming connection and will fall back to the next channel in their channel set."
BlazeDS Developer Guide, page 50.
This I did not expect, since non-blocking IO has been part of Java for over 5 years in the NIO package. If I were to deploy BlaseDS on an Intel sized box, I suspect the blocking IO would limit me to maybe 300 concurrent users. If this is true, it means that both streaming and long polling are unusable for almost any application on the public web.
Maybe the only option is to write a server in Java NIO that communicates with Flash Sockets.
Did I miss something?

A few questions about audio player in iBooks Author

1. How can I set the player to a small icon in Landscape layout? It seems that I have to display the player in the Landscape layout and show a big icon (which can't be resized) in the sidebar of the Portrait layout. Is there a way to customize this?
2. I notice that the audio player in iBooks Author looks different from that in the actual iBook on iPad?
In iBooks Author it's silver, which looks pretty nice:
But in actual iBook on iPad it becomes black:
Is it possible to customize the look of the player?
3. Another problem about the player in the actual iBook on iPad: When I have multiple players on the same page, only one of them shows the progress bar and time; others just show a play button:
If I click the one which only has a play button, the whole player will disappear (and won't play the audio clip). I must click the area (which the player is supposed to be in) again, and the player will show up with the progress bar and time (and other players will only have a play button).
This is very annoying. Does anyone have the same problem?

assuming you have a MovieClip mc on stage with the registrationpoint in the center:
stage.addEventListener(MouseEvent.CLICK, zoomHandler);
var currentZoom:Number = 1;
var zoomFactor:Number = 1.1;
var maxZoom:Number = 2;
function zoomHandler(e:MouseEvent):void{
    var zoomCenter:Point = new Point(e.stageX, e.stageY);
    var mcCenter:Point = new Point(mc.x, mc.y);
    var _xCorrection:Number = mcCenter.x-zoomCenter.x;
    var _yCorrection:Number = mcCenter.y-zoomCenter.y;
currentZoom +=zoomFactor-1;
if(currentZoom<maxZoom){
    mc.scaleX +=zoomFactor-1;
    mc.scaleY +=zoomFactor-1;
    mc.x -=_xCorrection*zoomFactor*.5;
    mc.y -=_yCorrection*zoomFactor*.5;
else{
trace("maximal zoom level reached");

Question about Audio line-in/optical in/Audio out/optical out

I have a late 2009 27" iMac (v3.06 GHz Intel Core 2 Duo). I note that on Apple's description of the rear ports, what I thought were simply headphone (Audio Out) and microphone (Audio In) jacks are described as:
1. Headphone/optical digital [S/PDIF] audio out:
"Line/Headphone output
"The headphone output is automatically selected for audio output if no external S/PDIF optical digital output device is detected. The headphone output supports a stereo data stream at bit depths of 16, 20, or 24 bits per sample and at sample rates of 44.1 kHz, 48 kHz, or 96 kHz. The headphone output volume can be adjusted from 0.0 dB to -64 dB..."
"The headphone output is automatically selected for audio output if no external S/PDIF optical digital output device is detected. The headphone output supports a stereo data stream at bit depths of 16, 20, or 24 bits per sample and at sample rates of 44.1 kHz, 48 kHz, or 96 kHz. The headphone output volume can be adjusted from 0.0 dB to -64 dB."
and
2. Audio line-in/optical in:
"Line input
"The analog line input operates independently from all other audio input ports and is always available. The line input supports recording at bit depths of 16, 20, or 24 bits per sample and at sample rates of 44.1 kHz, 48 kHz, or 96 kHz. Audio recorded from the line input is presented as a stereo data stream. You can adjust the line input gain from -16 dB to +30 dB..."
"The digital audio input has the following electrical characteristics (nominal specifications), based on input of a 1 kHz sine wave at 0 dBFS input level, 24-bit sample depth, and 44.1 kHz sample rate (unless otherwise specified below):
FSI – input sample rates: 44.1 kHz, 48 kHz, or 96 kHz
Bits per sample: 16, 20, or 24
Signal-to-noise ratio (SNR): >130 dB
Total harmonic distortion + noise (THD+N): <-130 dB (0.0003%)
Sound output/digital audio output port"
While, as an Olde School audiophile guy (62, and I still have vinyl!), I generally understand the gist of all this, I'm not at all hip to what S/PDIF optical digital output and/or digital audio input can do for me. So my questions are:
1. Can somebody please wise me up to the usage(s) of these optical inputs and outputs on the iMac?
2. The Apple description says,under "Line/Headphone output":
"PCM: 16, 20, or 24 bits per sample at sample rates of 44.1 kHz, 48 kHz, or 96 kHz
AC-3: 16 bits per sample at sample rates of 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, or 192 kHz"
and under "Line input
"The analog line input operates independently from all other audio input ports and is always available. The line input supports recording at bit depths of 16, 20, or 24 bits per sample and at sample rates of 44.1 kHz, 48 kHz, or 96 kHz. Audio recorded from the line input is presented as a stereo data stream. You can adjust the line input gain from -16 dB to +30 dB."
How -- where -- do I adjust these bit depth and sample rates?
Thanks for your help!
Bart

Thanks for the info. Turns out there's an even easier way to do it than using the "Audio Midi Setup": through the prefs in the latest (and probably some previous) version of VLC, which is currently Version 1.0.2 Goldeneye (Intel 64bit).
Now if we could only get full native resolution (or at least 1080p) HDMI input into the Mini-DPs of our 2560 x 1440-capable 27" iMacs and actually have it display something WAY better than 720p (half the resolution of which the 27" iMac is capable) to which Apple limited it, without having to buy a third-party $270.00 HDMI-to Mini-DPP converter and scaler. Call me crazy, but it seems kinda silly to have to pay another $300 to buy a device that will scale the 27" iMac's resolution back up to a measly 1080p, let alone its native 2560 x 1440. But, as the
As a side note, can anyone tell me -- perhaps the elusive and chimerical, but omniscient and omnipotent "communityhosts" of Apple.com -- why, in my original post in this thread, the phrase "1. Headphone/optical digital S/PDIF audio out" had the acronym "S/PDIF" stricken through, like it was a particularly egregious Anglo-Saxonism? Wikipedia defines the acronym thus: "S/PDIF is a data link layer protocol and a set of physical layer specifications for carrying digital audio signals between devices and components over either optical or electrical cable. The name stands for Sony/Philips Digital Interconnect Format."
If I have inadvertently broken another arcane, abstruse, and/or inscrutable canon of the rules (http://discussions.apple.com/help.jspa) governing what may or not be asked or said on these avowedly "user-driven" discussions, allow me to beg for forgiveness beforehand, pleading ignorance, and ask that Apple's "community hosts" or administrators would expand upon and make very clear, in their "Apple Discussions Terms of Use," a comprehensive list of terms, subjects, lines of discussion, references to certain items, products, people, policies, etc. that are simply disallowed in the forum.
It is extremely depressing and frustrating to spend however long it takes -- and it takes ME a long time, as I have a 60-year-ingrained tendency towards verbosity -- to ask and post a question or message, only to have the message never even posted to the forum, but, without any sort of consultation with me so that I may learn from, and correct, my errors, uncaringly consigned to the illimitable void by hands to whom there is no appeal or orison.
It seems unfair.

Two questions about Audio Tracks

I need to make several adjustments to an audio track of an interview to remove the voice of the interviewer who would give the occasional "uh-huh" etc., sometimes overlapping (though usually not) with the interviewee.
The way I've been doing this is to pull down the db level at those points in the timeline.
My questions:
1. Is this the best way to do it?
2. Regardless of the answer to question 1, why is it that sometimes when I use the pen tool in one stereo track, it will automatically affect the other at the same time, and other times I need to do each stereo track separately?
Thanks a lot.

If you recorded with two mikes, which it sounds like you did, you really recorded two distinct tracks, and you ought to bring them in as dual mono tracks, rather than stereo. (A capture setting.) Go to the Modify pulldown and uncheck "stereo pair." Then you should be able to adjust to your heart's content.
Did you record any room tone? It's very useful to fill in the gaps you create by eliminating words. Nothing detracts from an otherwise compelling show like bad audio! If you don't have any room tone, look for a section where no one is saying anything and there is no clothing rustle, etc, FROM THE TRACK where you are making deletions. Use that to fill the holes.

I have questions about audio timing and volume levels

I have a question regarding editing the audio timing. There are a few section in my clip that the rythym is a split second off. What technique can I use to sync up the time by just a fraction of a second?
so far I've tried use the re-time editor which I thought was gonna work but after I re-timed the section, I couldn't drop it back into place without having the lower clip change and make things more confusing, but I had a feeling I was doing thing right for a second. Was I on the right track or is there another way of doing this?
on another note...
I have some sections in my clip that are peaking in the red, but sound fine on my computer even at full volume. I hear no distortion at all. Does this pose as a problem, and should I re-adjust?
how to connect cuts in final cut pro

This is actually a somewhat complicated subject. There are actually several different kinds of sync issues. Two major "classes": 1) some audio seems to speed up or slow down over time compared to other audio, and 2) audio that just needs to be aligned in time.
One of the biggest culprits for sync issues is mixed bitrates. A typical FCPX project should default to an audio bitrate of 48kHz. If you have 44.1kHz (or other bitrate) clips, mixed in, the first thing you should do is make sure all your audio clips are conformed to your project.
Select your clips and then select:
[Note: when retiming clips, even if only slightly, make sure Preserve Pitch is selected.]
If that doesn't work, or your audio is "live" recordings and you'd like to tighten up performers timing a little, you can manually sync audio tracks to each other. (You should have at least one audio "track" with the best timing to be your "beat track".)
The idea is to find the events you need sync'd in the audio waveforms; which means listening and seeing the pattern in the waveforms (turning on "scrubbing" helps); and you should find these indicators as close to the end of the clip as possible (you will need to zoom into the clips to see them clearly enough and you will need room to access the retiming slider on one of the clips... so... near the end is best.)
Zoom into the audio clip on the storyline so you can see the waveform clearly, but not so far as to lose detail in the "subsample" range -- to start (it's very difficult to distinguish specific peaks if you zoom in too far.)
With the audio clip selected and the playhead placed on the start of the "event" (bass beat, tom/snare hit, cymbal, etc.) type the M key to set a marker (markers on audio clips can be set on subsamples, so it will be "inside" a single video frame range.)
On corresponing clips that you want to sync with the "master" clip, do the same thing: select the clip, find the corresponding "event" start for that clip, move the playhead to the beginning of the event (you might have to zoom in [command- +] on the storyline one or two times more to improve the resolution), type M to set a marker.
You now have two markers you can align by retiming one (or more) of the clps to match the "master". Use the playhead and have it 'snap' to the master's marker. When you retime the others, visually align their markers to the master's by using the playhead as a visual indicator for alignment.
This technique can be used with video and audio that has gone out of sync, by finding the visual indicators of a specific sound in the video, setting a marker to mark the start of that event, then finding the audio event and setting a marker. You will need to detach the audio from the video first if you're going to work on it in this way.
It's even easier to sync audio if there are no retiming issues (that is, you just need to align clips, not retime them.) Find the position markers as in the description above (anywhere in a clip since retiming is not necessary), turn on snapping, drag one of the clips until the markers snap... and you're done. You don't even need to use the playhead to have the markers snap to alignment with each other. Very convenient for adding foley/sound effects to video.
HTH

A question about project settings..FCP 7

I am using FCP 7 and I'm a little overwhelmed with the myriad of project settings options.
I use a Canon T3i and film in full HD (1920x1080 i think). All my footage I transcode in compressor to Apple ProRes 422 before importing to FCP projects.
I installed another computer and reinstalled FCP 7 and lost my old templates and importing some footage, I get the red line indicating unrendered footage.
Is there a standard setting that Compressor and FCP use to make it easy to just transcode, import, and edit? I'd like all my projects to look the best possible in full HD and there's hundreds of preset project settings and I'm a little unfamiliar with all the options.
Anyone have any quick tips, pointers, or thoughts?

If you drop a clip into the timeline, it means the sequence settings don't match the clip settings. Make a new sequence, add the clip. FCP should ask "do you want the sequence to match the clip settings?" Click YES. If you don't see that, you might need to trash your preferences...or it might be an older version that doesn't support ProRes formats (FCP 5.1 and earlier).
Another way is to choose an EASY SETUP that matches your clip settings, and then make a new sequence.
And your camera shoots 1920x1080p...not i.

Question about audio in and xbox360 audio on the x-fi audio c

I am having an extremely difficult time getting my audio to go from my 360 through the card. I have updated everything and still can't get any audio. I actually had audio for a day or two and then it stopped and I have no clue why it started or stopped.
I have my 360 hooked up to my monitor through the vga adapter and the audio is plugged into an adapter that goes to a single cord that plugs into the back of the card.
I had this previous settup under windows xp on another motherboard and the audio worked fine. I installed vista on a new mb and the audio wouldn't work with onboard so I bought the card. I get the same issue with the card although it did work for a few days as I said. Then it just stopped working and I hadn't touched any audio setting directly.
I've gone through all the audio setting I can find and have tried switch things on and off under sound which lists speakers and digital audio interface under playback. The audio from winamp goes through the digital audio interface (I can see the sound bar going up and down on it.
There's another tab that says recording and includes s/pdif-in, line-in auxiliary and microphone. All of those are unavailable except for the microphone. I've tried switching to the other devices and none of them have made the 360 audio come out.
I've also checked the hardware configuration under hardware manager and the device is working fine. I'm sure it has something to do with vista because everything worked under xp before.
Anyone else have this issue before? It kind of sucks that I can't play audio from my 360 to my speakers. I'm looking at installing linux on my other partition and see if it works through there....

Yep. I have the latest audio drivers for the card for vista and it was working. I was pissed because I couldn't get the audio working, then all of a sudden it worked and I have no clue what I might have done to fix that. I was playing bioshock for a few days and then I went one day to play it again, turn on my 360, and no audio at all.
It has to be either some audio setting, a driver issue, or some sort of incompatability in the software. Vista says the device is working and I do get audio out to my speaks from the computer so it has to be an issue specifically with the settings for the I/O jack, I think.
Like I said I had the same issue with my onboard sound so I bought the card. So i'm guessing it's some sort of vista setting. I'm just wondering what to check in the vista sound menu's. I have only just started using vista so I am not completely up to date on vista. The sound menu seems far more robust in terms of devices displayed and such.

Bursting - question about delivery channel

Can I call scripts with one of the delivery channels? Or maybe I can do it through filter, something like
<xapi:filesystem output = "/tmp/1.pdf" filter="/tmp/myscript.sh /tmp/1.pdf">
??

Many reasons. For exampel one of our request is to call web-services for some bursted file. If I could call a script this will become doable.
You mentioned BurstingStatusListener . I didnt' find any reference to BurstingStatusListener in Administration guide. Can you give some example?

Question about Logic Channel EQ

When i use Logic Channel EQ to cut low frequencies of say, a kickdrum (in this particular case up to 200hz), the level meter on the channel increases quite a bit. I thought cutting frequencies is supposed to reduce the volume? Also there arent any other plug ins on the channel. Is this normal?

At it's default setting the low cut on the channel eq has quite a bit of filter resonance at the cutoff point. This resonant boost is what you are hearing. If you lower the Q all the way this resonance will be gone and you will get only the cut you are looking for. If you use the graphic analyzer be sure to set it to maximum resolution as it is quite misleading otherwise.

Question about audio channels in FCP

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