Fcp X share export media export video & audio but I simply can't work with audio

I simply wanna (and even tried to) save a movie using whichever codec, I make sure to set it up on "export video & audio", but I cannot do anything on audio since the audio file format won't be highlighted (just like if there was no sound on a video & audio file !!) It works if I choose an "audio only" file, but that's not what I want - I want to export video & audio. !!!
Is it a matter of codec or a preference setting... I just cant figure out !
Has anybody ever got the same trouble?
Cheers

It was set to Surround and 48Khz. I set it to Stereo (and 44.1Khz for good measure, as that is the sample rate of my soundtrack) and then re-exported to H.264, and now the audo is perfectly centered.
So that's perfect, thanks.
Now the only thing wrong with FCPX is how it crashes every ten minutes or so!

Similar Messages

Working with audio | Learn Premiere Pro CS6 | Adobe TV

This movie introduces the robust audio tools in Premiere Pro. Abba Shapiro shows how to preview and trim audio in the Source Monitor. Whether it’s an audio-only file or the audio channel of a video clip, Abba shows how to enable sub-frame audio scrubbing and trimming. He also shows how to interpret clips to combine or separate stereo channels. At the same time, he introduces fundamental audio tools that are essential to working with audio in Premiere Pro.
http://adobe.ly/To00IZ

Is it possible to specify what audio channels to capture during capture from tape? It seems Premiere wants to mix the lavaliere mic and camera mic into a stereo track all the time, and then I have to modify audio settings and force it to mono in the project. In Final Cut pro 7, I am able to specify which audio tracks to capture, and this simplifies things in editing. I cannot see anywhere to do this in Premiere....

Bugged audio playback? When I start working with more than 1 audio track.

Hi!
I have been using premiere elements 12 for editing FullHD videos.
Brand new for me just bought it like 2 weeks ago.
Now during the last couple weeks I have had this MAJOR proglem with editing the audio.
This problem is headache and turns my simple editing into almost impossibilites.
I have had this problem with all my projects. _ALL_
Usually the problem seems to appear when I want to start working with several audio tracks.
At somepoint my projects just get ruined, I simply cannot hear the audio tracks.
The audio starts to disapear randomly from all the audio tracks.
It is not only playback problem. When I exported the video, the audio was partly missing, same as the project.
I have tried to look help on forums and search all what I could find, but havent been able to solve this.
Small fix has slightly helped:
I export audio to single track and bring it back to project. This works but I still usually have more editing to do.
I simply cannot understand why the playback disapears, I fear it is a bug.
I found lot of earlier version problems on people with similar type of problems, but no help for this version of program.
To solve the matter I have changed my audio setting in PE12.
Tried Premiere Elements WDM Sound and ASIO4ALL V2, but same problem on both.
I have looked up that my audio playback device and PE12 matches, and are not muted, and volumes maxed.
I have deleted my C:\ appdata\roaming\adobe premiere elements & premiere elements 12 folder preferences.. not helping..
I press the update pretty much everyday hoping for new version.. But yes, program keeps telling me its the latest.
I usually only work with 3 audio track maxinum. And I am doing really the most basic stuff here..
I have set all my audios to foreground and it has given me best results, but it does not seem to take the problem away as at somepoint randomly my audio playback is missing again, randomly across the playback.
Please anyone? I am feeling hopeless here.
Info on my computer:
Operating system: Win7
Processor: AMD Phenom(tm) x4 955 processor 3.20 GHz
memory (RAM): 4GB
system: 64bit
<moved from downloading, installing, setting up forum by mod- kglad>
Message was edited by: kglad

First of all, thank you for replying my message!
I will go and try to answer best I can.
Source media properties:
Progressive frame rate, I record this from a game.
Recording program: dxtory
codec: Lagarith Lossless codec
Format: avi
Frame width: 1920
Frame height: 1080
Data rate: 1990656kbps
total bit rate 1992192kbps
Frame rate 30/frames/seccond
Audio of video properties:
Codec: PCM
Format: 16bit, stereo
Bit rate: 1536kbps
Channels: 2(stereo)
Audio sample rate 48 kHz
In addition to that I use wav files for music and narrate
Music files sizes 20-50mb
bit rate: ~1411kbps
In total sizes of audio involved is usually something like 300mb
What are you or the program setting as the project preset in the new project dialog. See Edit Menu/Project Settings/General and its Editing Mode, Timebase, and Frame Size readings.
I opened new project and this is what i got:
Editing mode: HD 1080i
Timebase: 25,00 frames/second
Frame size: 1920 x1080
Audio is 48000 Hz
(I dont know how to change these, btw. Program don't seem to let me change editing mode.
I also looked up my most recent project and found these settings:
Editing Mode: DSLR
Timebase: 30,00 frames/second
Frame size: 1920 x1080
..and same audio settings
Do the problems persist if you start a new project? Do you find the audio problems develop when the Timeline content is fairly well developed or can the problems occur when there a minimal amount of content of those tracks? Let us look at the size and duration of this project.
Yes, I have had this problem with all my projects. Sometimes it goes away tho, just enough for me to work bit more. My work progress usually starts by clipping the video material, like 1h material to 8minutes in clips. After that I want to work with audio tracks, and here problems start! When I start working with more than 1 track, so the problem is at later stage. To fix the problem a small fix I have used, I export audio like music and effects tracks together to 1 wav and bring it back. After that, I start working with narrate track. To describe the problem with audio:
Audio Randomly starts to disapear when I playback, and can also be gone if I export the video out.
My source videos total size are usually within 80Gb max limit and audio edit material 300mb total.
In my last project my biggest source video was 47GB and leingth of 1h 4m, in addition to that I had 3 smaller videos, sizes of 1GB, 1,54GB and 300MB.
Is there any evidence that you are accidentially moving the audio track rubberband to alter the audio volume. That rubberband is the orange line that runs horizontally through the middle of the audio clip when you are looking at the wave form display style of that audio track. The rubberband represents audio volume.
No, audio volumes might have been adjusted in my tracks but they are present. The problem is usually like ~2-5seconds, a random audio track can be heard (like music or narrate) or just no sounds at all. This makes it very difficult to work with the audio. Also I gotta say that this problem starts by simply having the 2-3 audio tracks.. In my recent project the only adjustment I used to edit the audio was fade in/out.
I don't have any other audio problems with any other programs.
I am more than happy to provide more information if needed. I hope this information helps.
In addition I can say that when I close Adobe Premiere Elements 12, it crashes. No idea why.
I have also looked up my firewall settings, anything involved adobe and premiere..
In addition some effects like HD text templates make "Adobe Media Core CS6" crash. I tend to just not use em, or if I need em, I add em at end of the edits.
I hope this helps, I would really like to solve this problem.
Thank you alot for taking the time to answer! I welcome anyone to help mee too! Please giev links to similar problems too. Thank you everybody!

Working with audio

Hi,
I've got a couple upcoming projects that will largely be focussed around music, and animating photography in AE in time to the music. I've checked the help guide, but can't seem to get preview with audio functioning properly. Is this even possible in AE?
At the moment, all I can do is preview sound "from this point onwards" using numpad "." but I need to see the video track simutaneously, in order to fine tune the animations. What's the best way to do this?
Thanks,
Danny

Yes it is possible, you can work with audio and you can play it back of course, use the RAM preview button.
But, AE is not the best program to work on Audio with. In fact most people I know and myself, avoid using Audio in AE at all costs. Now it's case of what you like, some don't mind audio in AE, but for most, the response times and additional features when working with multiple layers of audio just aren't good enough.

Unexpected problem: uncaught exception when working with audio files in Soundtrack Pro

Hey,
Whenever I double click an audio file in the timeline to work on it, I end up getting the Error message: "Unexpected problem: uncaught exception" and Soundtrack Pro eventually crashes. Also, this happens when I try to work with audio files in the file editor. I have Leopard and am using Logic Studio 8. Any insight would be much appreciated. Thanks
Matt

Didnt know how to remove the original clip from the STP arrange window - or whatever it's called in STPro. So I could go no further in my perverse science experiment.
It also appeared that I could only import an aif but not a wav?
I gave up on STPro and did what I needed to do in Logic, which I use regularly.
Thank you for folowing up on this.
PS. I used to use an old PowerPC version (1.0.3) of STPro and loved the way it visually showed the change in the waveform when you added each efffect. It appears that the new version (3.0.1) does not allow you to do this anymore? You can render or flatten the effects to see the final waveform, but then you are dead in the water and cant make changes?
Im such a newbie at STPro that I am probably not using it right?

Why the Labview2012 can't contact with Audio precision SYS2722, who know that?

I intalled the Labview2012 into my laptop, it can work well with audio precision APx585 through APx500 drivers, but it can't work with AP SYS2722 even I copy the driver to NI installation directory. Howevery, I can use the AP's control softwrae(client) properly.
Anyone can help me?

I used both to call to them
Is it ok if I will send the adapter back to apple shipping without permitting from call centre?

Working with Audio Premiere as well as Exporting to right codec

Hey guys,
Before I explain my problem let me tell you my setup. I have 2 Mac Pros. One is the new 12 core with 20 gb of ram(snow leopard) and 1 Early 2008 8-core Mac Pro with 20 gb ram(snow leoaprd). One the 12 core machine I am using After Effects CS5 to which I do my composting I render the file as the default animation codec with audio. I import the footage into Premiere CS5, which is on the 8 core early 2008 mac pro which has a 1gb ATI card so not Mercury engine capable, He has problems with it crashing when trying to keyframe and applying effects to the audio. Also as a test we render a sequence and there are drops in audio when in fact there are none in the sequence. What could be causing this? This is concerning because this small 9 minute sequence is only part of a short kids movie for our church which we shot in HDV 720p footage via a Sony FX1000. There will be many sequences that will make up about a hour and 15 minute movie. Should we render the sequences that make up the movie with the changes in audio, Import back into premiere that then piece them together with transitions to put out the final product, or should we, which I was thinking we could do is just use the sequences as they are and put them together? Or this why we are having so many problems? Sorry for the long explanation. I just wanted to be thorough. The editor is new to premiere, he used speed edit by Video Toaster for years. I have used After Effects for a few years now.

What do you mean by a better routine?
A better compression routine than H.264.
Is the quality better than H.264?
Yes.
Does this use an MOV wrapper?
Yes.
Also, would it import correctly into PPro?
Yes.
we would import files into PPro exported out of FCP and would get a red pixelated glitch during preview and export. Does this codec solve this problem as well if it is an MOV wrapper?
I don't know anything about FCP.
Sorry for all the questions, just want to get things right before diving into a new codec.
It's harmless, and free.
Download it and run some tests.

Does FCP 4.5 have audio channel bug, and can only do 2 audio channels?

I transferred my digibeta project with 4 audio channels (2 are Music & Effects) to an uncompressed QT at a dubbing facility. I can play the QT with all 4 channels heard on my G-5 Powermac, but when I import the QT file into FCP 4.5 (to do some re-cutting), it will only import two audio channels (the Music & Effects) and not my other two stereo tracks with dialogue. and it's not the timeline because I have four channels available -- it's the actual imported file losses two of the audio channels on import into FCP. I tried importing the same file into iMovie and it plays all four channels,unlike FCP. I have three computers with 4.5, and each of them have the same problem. Is there a setting that will change this, or should I just throw FCP 4.5 away. Will I have the same problem with 6.0?
Bryan

If you want to work with 4 Channels, throw out version 4.5 and get 6. Version 4.5 could only work with 2 channels. This changed with the 5.0 version. It's not really so much of a bug with 4.5 as just not being implemented in that version.

How do I get my iChat to work with Audio/Video?

I have tried to read through the other posts on this topic to figure out what to do. I am quickly in over my head with changing configurations on ports and routers. It should not be this difficult. Until I bought this computer less than a week ago, I had used Windows machines which I would be able to figure out fairly quickly. I thought Apples were supposed to be easier to use? I have been working on this issue for two days now. I even called Apple twice. The first guy said what I was asking was beyond his knowledge, and I should log into this discussion board for an answer to my question. I told him I could have done that without paying for both Apple and Pro Care.
The second guy worked with me at least and put me on hold several times to research. He had me try different things before telling me that I must not have enough bandwith on my ISP. He told me to call my ISP provider and ask them what the bandwith was. When I asked him what bandwith I needed to iChat A/V, he said he could not tell me what it was. If it is a bandwith issue, shouldn't the screen say so? Instead it tells me that I failed to accept the invitation that I initiated, or failed to accept the invitation after I pressed the accept button.
I know the MBPs are new, and there will be growing pains, but I expected a little more assistence from Apple. I am working dilligently to learn how to use my new Mac and am getting completly frustrated with this A/V Chat problem. I even downloaded AIM on my MBP to see if that fixed the problem. The same thing happens. Either I initiate the video chat and the other person accepts, which causes my screen to say that I failed to accept the invitation. Or they initiate the invitation and once I accept my screen freezes saying that I failed to accept the invitation.
From the looks of the other posts, this does not seem to be an Intel chip issue. Since AIM did not work either, it must not be an iChat issue.
Thank you in advance for your help.
MacBook Pro Mac OS X (10.4.5)

Hi Kambini,
I think you have quoted function of the Linksys rather than the model number.
Is it one of the ones listed on this page http://portforward.com/routers.htm
11:41 PM Monday; February 27, 2006

Issues Working With Audio

Hi
Has anyone worked out something for these situations?
1: I have source footage with 2 tracks ... dual mono... wireless mic 1 and wireless mic 2. I edit the clip into a Project (set up as Stereo out) and need to balance the two audio clips. I can select the clip ... set it to Dual Mono in the inspector... and Pan Left/Right. But to adjust the levels of the individual channels it seems I have to right click on the clip in time line and select "Open In TimeLine". Now I see the 2 channels. The Format settings need to re-set to dual Mono and pan settings to Stereo Left / Right .... now I can Pan the tracks and set a Level Envelope curve for each on the timeline... but can't hear the rest of the Program Mix!!!
Is there a way to just get all of clips's audio tracks to show in the Main Timeline. Expand doesn't seem to do it? Seems soooo basic.
2: If Im editing video and audio into the timeline how do I choose which audio tracks to edit into the program. I see how to select Audio / Video or both .... but if I just want one track? Do I have to keep switching audio tracks on and off in the Source clip via the inspector?
Thanks
Lee

Sorry ... don't know how I posted here ... off to FCP X !

How can I sync with audio framerate?

I hope that there is an entirely different way of doing what I'm trying to do because I've run up against what seems like an insurmountable obstacle. I have been working on a project for many months and the last piece to figure out is synchronizing application actions with the audio framerate. I have seen other apps doing this successfully (but proabably written in C++, not Java) so it can be done with standard Wintel boxes - but I'm now concerned that it cannot be done using Java Sound.
My project attempts to animate sound with visual images (not video or pictures but more abstract visualizations based on the audio itself). It works fine when simply "listeing" and responding to the audio. The final step is to be able to sync specific abstract visualizations with precise points in the audio (frames or position in microseconds). I'm now wondering if what I need is handled by the soundcard (hardware) and not available through Java. I presumed that I would be able to rely on the soundcard to handle the realtime audio stream (using its clock) and I would simply wait for increments of a certain frame count that correspond to the refresh/trigger times for my visualizations. As it turns out, after a lot of experimenting with my configuration, I can't find a way to trigger a method in my application synchronized wiht a certain period in the audio being played. I am playing audio by writing to a SourceDataLine, and listening (using loopback, I believe) on a TargetDataLine. To give an idea of what I'm looking for, if a LineListener supported notification based on reaching a certain frame position in the audio stream, that would be great. That doesn't exist (that I could find, only START/STOP/OPEN/CLOSE events). I've been trying to set the buffer size on the TargetDataLine, and the number of bytes (frames) that I read from that TDL, to achieve a synchronization with the actual rate the audio is playing through the sound card. What I've seen is that the number of frames made available through TDL is not consistent. The read() method blocks for various times and with differing incremental frame positions as the TDL buffer is replenished in real time.
My goal is to be able to detect when a number of frames corresponding to 20 milliSeconds at 48kHz or 44.1kHz sample rate have been played through the sound card. Is this possible? I've not posted any code because the code I have isn't non-functional, it's just not capable of performing the task I am trying to achieve. Suggestions as to how I can refine my existing approach, or what other direction/implementation I can pursue will be greatly appreciated.
Edit: typo.

ags wrote:
I think this breaks down into two related but separate issues:
First, the latency between writing audio (to a SDL) and when that audio is actually heard. If I could find a way to measure that, and it was consistent, I could adjust my processing and delivery of audio data to that SDL. As I tried to explain in my previous post, there will ALWAYS be a variability in the time when a given frame is processed in an SDL and when it sounds. Biggest variability is between the time required for the onset of the first sound (played via bytecode interpretation) and onsets of subsequent sounds (if played via code loaded into memory). But JVM switching also accounts for variability.
Have you seen this article?
http://quod.lib.umich.edu/cgi/p/pod/dod-idx?c=icmc;idno=bbp2372.2007.131
"Real Time Low Latency Audio Processing in Java"
It does a good job of listing various factors that affect real-time audio latencies.
The other problem is that even if I were able to get an accurate measurement of the delay (latency) in the audio lines, if I then set system timers to cause action in sync with the audio frame rate based on audio sample rate (frame rate for uncompressed PCM audio) there is still an error path. Again from observation not direct knowledge, it seems that there is no direct link between the sound card clock (controlling real time frame rate and what should really be the master of all actions) and the system clock (which I would use as a surrogate for the actual sound card clock to synchronize my actions with the audio being heard). Even if I were to establish perfect synchronization at the start of a song, by the time 5 or 10 minutes of audio has been processed, I suspect there will be an accumulation of errors between the sound card clock and the system clock that becomes noticeable.The sound card clock and the system clock are perfectly in synch. The problem is the granularity of the system clock. With Microsoft, the clock info is only updated once every 15 msec or so. As a consequence of this, System.currentTimeMillis() will only report the time at the last update. Also, Thread.sleep(millis) and setting a Timer are subject to this constraint. The error doesn't accumulate, though. It is just an issue of having a reduced "granularity" that creates a predictable error. (System.nanoTime will always be more accurate--it is not subject to letting the OS determine the update frequency.)
As I wrote, there is a hack that allows you to increase the granularity of the OS clock interrupt. It involves setting an independant, background thread to sleep for the maximum amount of time, e.g. Thread.sleep(Long.Max_Value). This somehow forces the OS to use a higher granularity, and with Microsoft the accuracy becomes about 1 msec (comparable to Linux and Mac systems). I learned this at the forum Java-Gaming.org.
>
I'm open to ideas, criticism or examples of how I might acheive my goal. While it might not be practical or the only way, the best I can describe what I'm looking for is to have an audio-event driven system where I can determine a frame position corresponding to the audio being rendered in real time and trigger events to keep other actions synchronized with that audio. If a constant delay (latency) was maintained and measureable, that would be OK.
This is a difficult problem.
Agreed about the difficulty!
Given all this, I think the questions are:
1) how much in advance do you know about given frames that you wish to use as cues?
2) how well can you guestimate an anchor point? (and, is there a way to improve upon the guess with feedback?)
3) how accurately can you trigger the visual cue?
Regardless of when Java decides to process a frame, it DOES manage to keep the playback steady. Thus, if you focus your efforts on finding an ANCHOR (for example, the nanoTime, or "close enough" approximation, when the first frame sounds) that you can use as a reference point for the duration of the sound, then you can use that point to calculate conversions between the frame count of the sound and the time it can be expected to play.
For example, if the frame is 44100 and you estimate frame 0 was at time 1336426670003 (System.currentTimeMillis), then set a Timer, in advance, to trigger the visual cue at 1336426671003 (I'm adding 1000 msec). The level of accuracy will remain the same regardless of the frame number and duration involved as long as you use the same anchor (and the sound playback is not interrupted).

Why iPhone 3G Dock doesn't work with audio cable?

Hi, I'm Italian. I apologize for the mistakes, but I just learned English!
I will try to explain you my problem. I bought an audio-video cable for iPhone to link it at my stereo and to get better the HiFi System, I decided to buy an iPhone 3G or 3GS dock (my iPhone is a 3G one). But, when I linked all, the iPhone works with its speakers. Is possible to get working the iPhone like this:
iPhone > audio/video cable > stereo
with these links:
iPhone > dock > audio/video cable > stereo
Thank for your help.
P.S. The products are all original, they're these:
http://www.memorypuntoit.com/public/prodotti/982010171817a.jpg

Have you downloaded the Remote App? Wouldn't it be better for your iPhone's battery life to just let iTunes play the music over AirTunes? I'm kind of confused as to what you're trying to do.

Can't work with Macbook mini-dvi adapter to s-video out under Windows..??

hi..
My macbook can work well with mini-dvi adapter to tv out under mac os,but it can't work under windows os..
If anyone's using their Macbook and has any comments please let me know..
Thanks.
Richard

Yes, if you can get an s-video cable and plug that into the SCART adapter, that would be better than plugging in composite video.
Clearly, having an SCART connection is not going to "improve" the video going in, you're going to be limited by the lowest quality connection. An s-video connection will provide a higher (but probably not a lot) quality signal than composite video.
-Doug

Mac Pro RAID block size recommendations for working with audio in Logic Pro

I have recently ordered a Mac Pro and plan to do a RAID configuration across 3 HDD's
The RAID type i am going to do is a RAID 0 striped.
The computer is going to be used primarily for audio post production and working with 20+ 24-Bit audio files at any one time within a Logic project.
I want to know what is the best block size i should use when configuring the RAID.
I understand that using a higher block size is best for working with large files but do i need to do this in my case or will the default 32k block size be enough?
Thanks in advance

Use 64k. Things like databases like having 32k blocks because of all the small files. Audio files are pretty small even at 24-bit 192KHz. Go to 128k if all you are doing is streaming and no samples. But 20+ 24-bit is really not too large anyway considering most modern HDD's can stream 100MB/s off one spindle. You'll probably be fine regardless of the block size you choose. But most audio pro's choose 64k.

M-Audio Fast Track Pro doesn't work with OS X Mavericks Garageband or MainStage 3

I used to be able to use MainStage 3 and Garageband with my M-Audio FastTrack Pro with no problem. Then, when I upgraded to OS X.9 Mavericks, it doesn't work. Avid (who now supports the FastTrack Pro) recently came out with an updated driver that supposedly upgrades for OS X.9 Mavericks (http://avid.force.com/pkb/articles/en_US/Download/Fast-Track-Pro-Drivers), but even after downloading/installing the driver and updating the device, I still can't get it to work with Garageband or Main Stage 3.
It seems that the software sees the M-Audio FastTrack Pro USB, I even see an audio signal in both MainStage 3 and Garageband for my guitar or microphone input, but there is no output sound from the FastTrack Pro to my PA system other than the analog sound (no sim amps or vocal effects work).
Anybody been able to solve this problem?

I had the same problem (Fast Track Pro + Mavericks 10.9.2 > no output sound).
After trying several combinations, I found out an non logical solution, but it did work
What I did:
1) Download the most recent driver update 1.9.4 (http://avid.force.com/pkb/articles/en_US/Download/Fast-Track-Pro-Drivers)
2) Uninstall Fast Track Pro Drive and restart computer
3) After restar, do NOT install again. Just go to the System Preferences/Sound and set Fast Track as your in/output port
Hope it helps!

Fcp X share export media export video & audio but I simply can't work with audio

Similar Messages

Maybe you are looking for