Recorded audio ahead of physical recording

Hey guys. So this weekend we were in the studio working on a new song. I'm having a problem when recording audio. I noticed it when we started laying down an acoustic track. The acoustic is stereo mic'd and I'm recording on 2 separate mono tracks. I have low latency mode turned on and I hear no latency at all through the headphones. After I record the track (spot on with the drums), we play it back and the track is rushed. I have to nudge the tracks right about 6 times to get it lined up to where it was actually played. The same goes for every track I recorded, even to the point where I had to use flex time to line the lead tracks up properly. I know for a fact I didn't play it rushed like this.
In my preferences, it shows a round trip latency of 17.7 ms and the sample delay slider is set to -646 by default. I have a Presonus FP10 that I'm using. I have no effects turned on for the recorded track (minus POD Farm for the electric tracks). This happens all the time, Logic is moving the recorded audio forward in time. I have to fix this, because we'll be laying down bass and vocals this week. There are very few other plug ins running in the session (and this is all the sessions too). The only bus is a drum bus using a compressor with a 2:1 compression set. The only reverbs are on the overheads and high hat tracks.
Why is Logic shifting my recorded audio forward?

gnogtr wrote:
the sample delay slider is set to -646 by default.
Do you mean the recording delay slider? If so, I suspect that's why your tracks are playing back early. The default value should be zero unless the Presonus driver is reporting the delay to Logic and Logic, in response, is setting that value to -646. But as far as I'm aware, Logic doesn't even do that.
Try this test:
• start with a blank song with no plugins anywhere, PDC set to OFF, and software monitoring set to off (_don't leave any of those steps out_)
• set the recording delay slider to zero
• take a track (something that has drums or distinct rhythm to it) and place it on track 1, bar 2 (yes, bar 2)
• take patch cords and physically connect the stereo output of the interface back into inputs 1 and 2. No, you won't blow anything up by doing this.
• create a new audio track, assign it to inputs 1/2 and put it into record-ready
• go into record and record about 10 or 20 seconds of the track from bar 1
• take the track out of record and play back both tracks.
At this point, do they sound like they're flamming or flanging? Or do they sound tight playing together? If they sound tight then your basic problem was that recording delay setting of -646.
This doesn't mean you're out of the woods yet, but the test I outlined above will put you on the right track.

Similar Messages

Audio ahead of video

Hi,
I work with low resolution video and graphics. I don't anticipate working with higher resolution at all, I am not going to buy a digital videocamera. I take pictures with my digital still camera at 640x480. My screen resolution is set for the lowest I can use iMovie with, 1024x768 (stretched). I capture video with an XLR8 USB device which feeds video from my tv/vcr/VHS camcorder. The driver is from InterView and I capture with either BTV or InterVew Capture. (See my separate thread on resolutions)
(I managed to get iMovie to show the video window at 641x480 but I will have to check each time if I want to be sure I am set right (although it doesn't really matter.) It would be nice if there was a setting that I could keep it at... Yes, I can try to leave the window at that setting but I would prefer it be kept as a preference.)
Some of the material I want to deal with is VHS video, video from the tv, webcam video that I have laid off onto VHS. I want to edit and put that stuff on DVD. I have have figured out how to do most of what I want to do (and it looks fine when played back on the tv which is all I am asking for) but here is a problem I am not sure hot to work around:
The audio is ahead of the video. I feed the audio from my tv to my stereo and from there it goes into the computer. It ends up pretty close. I didn't notice it until I started using iMovie and iDVD; I've captured lots of video before on an older machine. Help in iMovie says I should "extract the video", save the project to Quicktime and see if that helps. Isn't that just incase the problem is a Quicktime bug or something? In my situation, as far as I can tell, the reason the audio is out of sync with the video is that the video is going in via a USB 1 device and the audio is getting into the computer faster via the audio-in port. While capturing (with BTV or InterView Capture) there is a bit of a delay anyway, the audio and video are behind what I am seeing on the tv. It never really bothered me before but it is much more noticeable now when I look at the results (It's not noticeable during the capture.)
I could try to select the audio track (after extracting it)and try to noodge it to the right spot (using the arrow key?) but I am skeptical. It is so hard to cue with iMovie, trying to synchronize audio would be an exercise in futility. I might be able to noodge it to the right of left by using the arrow key but it will be real hard to tell if it is synched up.
Which brings me to another problem. The only way I have ever been able to see waveforms is by extracting audio. Shouldn't I be able to see them by just checking the menu item Show Audio Waveforms? It never showed them until I extracted them into their own separate track. (One other feature I would LOVE to see is being able to hear the audio at various speeds. I would be able cue/synchronize much better being to hear when something starts. Better movement controls would help me alot too.)
Does anyone have any ideas on how I can make sure my audio is in sync in the future? My XLR8 device does not have an audio input which, I think, would solve it.
Thanks.
John L

OK. Here is what I did for this round which wasn't successful (was still out of sync) but I wanted to try it anyway. At one point when I looked at the iDVD preview some of it actually seemed to be in sync.
Back to the beginning. I set iMove to 640x480. I get those numbers in yellow on the screen. (I had set the screen to 640x480 once before. It did not remember that when I started up iMovie...) I hope those numbers go away; they're sticking around...
I check the source files and the audio is good there. In sync. They are InterView Exported from the original QT format to .dv to get rid of InterView's encoding which supposedly prevents people without it from seeing the video (but I have never noticed that it made any difference. It does fix the 16x9 aspect ratio problem I noted in an earlier thread though so I do it.)
I import/save as MPEG-4. Importing slows down on the last two clips. It says it will be done in 23 minutes for a long time. Tabbing between a few other open apps seems to encourage it to speed up... I check Activity Monitor and there are no stalled processes. OK.
I have 13 GB available. I know I should have more but unless it gives me an error message I will attempt to go with what I have. (I did throw out all earlier versions of iMovie and iDVD files.)
OK, It's done importing. The 640x480 lettering has gone away, thank you. Now I will edit each clip separately. I think it will be easier that way. Hopefully putting them back on the clip tray(?) after I I'm done won't be a problem.
I check, the audio is NOT in sync but I will try to fix that with the suggestions from the link above, before I export to iDVD.
Trying to edit is very difficult. All I want to do is just cut some of the beginning and ends off of these clips. Sometimes I get a vertical bar with the arrow facing in, sometimes I don't. Also, sometimes it shows me the whole clip as being selected, sometimes it only shows from the beginning to a little past the playhead marker. Very weird (but alas, not surprising). Saving and quitting and restarting iMovie does not help. Eventually the vertical bars appear when needed and I move on. (I assume all I have to do is move that bar to the where I have left the playhead. That seems to work most of the time. I bookmarked certain points as well and dragged the vertical bars to those points and if it did what I wanted, I moved on. Sometimes weirder things happened but I can't describe them.
Now to put all the clips on the timeline and see if I can sync up the audio. I (Select all and) extract. The progress bars do their thing. I get a red bar under the first clip so I scroll to the right and eventually get all the of them done and proper waveforms. I extract. The audio is NOT in sync. OK. I try suggestion #3 (putting in one or two second gaps at chapter markers but have difficulty with that and give up. I don't have QT Pro so option #4 is out. Option #2 I discard since I want all my segments as one movie and as individual segments but I later realize I could do that anyway. My DVD would end up in slightly different format but it may be workable. I may try that if what I am doing now in iDVD doesn't work out. (I may just give up at this point too. It isn't worth the trouble...)
One reason I gave up with #3 was that I could not select all! I get the selected blue indication from the beginning and a little after the playhead but not after that. Other weirdness occurs...
I ignore that and put in chapter markers. I put in 4 for one clip and I see that the entire timeline is selected. Good.
I bring it into iDVD. (I suppose I should continue this thread in an iDVD message...) I'll just leave it here: things look out of sync in iDVD as I am encoding it. I doubt it will be fixed but I will let it run its course. I do have an earlier attempt which may be a little better/not bad.
OK. iDVD is all set up but the sync isn't any better. I just wanted to try it and see.
Maybe I'll try iMovie 3...
Thanks for your interest and all you help.
John

Audio ahead of video in iTunes using Windows 8

I recently bought a new HP ENVY m4 laptop with Windows 8 at Best Buy. I do not like Windows 8, I prefer Windows 7, but there was nothing I could do about it. After I bought my new laptop, I downloaded iTunes along with everything I have ever bought on iTunes. A few days ago I bought and downloaded the movie "Oblivion", the new Olga Kurylenko movie. I bought the Hi Def version of the movie to be specific. Every time I've tried watching it, about half way through the movie, the audio gets ahead of the video. I have to constantly pause then play the movie, to keep the audio inline with the video. This has also happened with some of the other movies and series I have, that are in Hi Def as well. The movies and series that I have in the "Standard/ Non-Hi Def" versions play just fine, with no problems. I was just wondering if anyone else has ever had this problem, not just with the same laptop I have, but with any other laptop or computer (Be it a Mac or PC). And if you have had or still have this problem, how can you fix this problem? It's starting to really bug me. As for my laptop; it's an HP ENVY m4-1115dx Notebook with Windows 8 64bit OS, an Intel 3rd generation Core i7-3632QM Processor @ 2.20GHz 2.20GHz with Turbo Boost Technology up to 3.2GHz, it has 8GB DDR3 SDRAM (2 DIMM) RAM, a 1TB 5400RPM hard drive, Intel HD graphics 4000 with up to 1664MB total graphics memory, with the Beats Audio, with a whole other bunch of goodies. I've talked to my local Best Buy "Geek Squad" and they've told me that there shouldn't be a problem with the audio getting ahead of the video, or even the other way around, because my laptop is set up to accommodate and handle Hi Def movies, thus the Intel 4000 HD graphics card, and also because my graphics card is a Hi Def graphics and audio card (But I don't really believe my graphics card is also an audio card, but I could be wrong). Now, I am no computer wiz or anything like that. I just watch and download movies, listen to and download music (I only download iTunes movies and music, I've learned the hard way not to download music and movies on P2P sites like Limewire and Torrent sites), surf the internet, email, and do work stuff, and stuff like that. I have some what good knowledge of computers and how they work, but like I said I am no computer wiz. So if anyone has any advice or help, I would really appreciate it. Sorry about the novel, I didn't mean to get so in depth with everything. Thanks again for any help.

I could never tell if it was out of sync in the video preview on my lap top because it was so jerky.
So the only way for me to ensure that it was in sync was to write it out to a DVD disk and then view the content on a TV.
(I know that it automatically transcoded it to be standard def instead of hi def)
And after doing that, I did not see any sync issues.
I know this method is too time consuming, which is why I build a desk top PC so I could do real time previews while editing.
I will try to write it out to a Blu Ray disk both from my laptop and from my new desktop PC to see if they both have audio sync issues.
This is very confounding.
I have also ordered a 6 pin to 4 pin 1394 cable (since my Canon XHA1 has 4 pin iLink firewire and my desktop pc has 6 pin) so that I can try to recapture to see if that is the issue.

Assistance with Recording Audio directly from my laptop

I have logic pro and I use a audio interface (M Audio firewire 1814). I have two synths and a mpc hooked up and working well. If I am listening to something off the internet or sounds from my internal soundcard and want to record the audio, what input information do I need to use? I can see activity in my interface 1/2 sw rtn window and hear the audio but can not find out how to record the audio. Thank you for any assistance you can provide!

Hi
I guess that you could use the built-in audio for "system and general" audio playback, and physically cable the output to the audio interface that Logic is using, but it might be simple to use a utility such as SoundFlower:
http://cycling74.com/products/soundflower/
CCT

A quick primer on audio drivers, devices, and latency

This information has come from Durin, Adobe staffer:
Hi everyone,
A common question that comes up in these forums over and over has to do with recording latency, audio drivers, and device formats. I'm going to provide a brief overview of the different types of devices, how they interface with the computer and Audition, and steps to maximize performance and minimize the latency inherent in computer audio.
First, a few definitions:
Monitoring: listening to existing audio while simultaneously recording new audio.
Sample: The value of each individual bit of audio digitized by the audio device. Typically, the audio device measures the incoming signal 44,100 or 48,000 times every second.
Buffer Size: The "bucket" where samples are placed before being passed to the destination. An audio application will collect a buffers-worth of samples before feeding it to the audio device for playback. An audio device will collect a buffers-worth of samples before feeding it to the audio device when recording. Buffers are typically measured in Samples (command values being 64, 128, 512, 1024, 2048...) or milliseconds which is simply a calculation based on the device sample rate and buffer size.
Latency: The time span that occurs between providing an input signal into an audio device (through a microphone, keyboard, guitar input, etc) and when each buffers-worth of that signal is provided to the audio application. It also refers to the other direction, where the output audio signal is sent from the audio application to the audio device for playback. When recording while monitoring, the overall perceived latency can often be double the device buffer size.
ASIO, MME, CoreAudio: These are audio driver models, which simply specify the manner in which an audio application and audio device communicate. Apple Mac systems use CoreAudio almost exclusively which provides for low buffer sizes and the ability to mix and match different devices (called an Aggregate Device.) MME and ASIO are mostly Windows-exclusive driver models, and provide different methods of communicating between application and device. MME drivers allow the operating system itself to act as a go-between and are generally slower as they rely upon higher buffer sizes and have to pass through multiple processes on the computer before being sent to the audio device. ASIO drivers provide an audio application direct communication with the hardware, bypassing the operating system. This allows for much lower latency while being limited in an applications ability to access multiple devices simultaneously, or share a device channel with another application.
Dropouts: Missing audio data as a result of being unable to process an audio stream fast enough to keep up with the buffer size. Generally, dropouts occur when an audio application cannot process effects and mix tracks together quickly enough to fill the device buffer, or when the audio device is trying to send audio data to the application more quickly than it can handle it. (Remember when Lucy and Ethel were working at the chocolate factory and the machine sped up to the point where they were dropping chocolates all over the place? Pretend the chocolates were samples, Lucy and Ethel were the audio application, and the chocolate machine is the audio device/driver, and you'll have a pretty good visualization of how this works.)
Typically, latency is not a problem if you're simply playing back existing audio (you might experience a very slight delay between pressing PLAY and when audio is heard through your speakers) or recording to disk without monitoring existing audio tracks since precise timing is not crucial in these conditions. However, when trying to play along with a drum track, or sing a harmony to an existing track, or overdub narration to a video, latency becomes a factor since our ears are far more sensitive to timing issues than our other senses.   If a bass guitar track is not precisely aligned with the drums, it quickly sounds sloppy. Therefore, we need to attempt to reduce latency as much as possible for these situations. If we simply set our Buffer Size parameter as low as it will go, we're likely to experience dropouts - especially if we have some tracks configured with audio effects which require additional processing and contribute their own latency to the chain. Dropouts are annoying but not destructive during playback, but if dropouts occur on the recording stream, it means you're losing data and your recording will never sound right - the data is simply lost.   Obviously, this is not good.
Latency under 40ms is generally considered within the range of reasonable for recording. Some folks can hear even this and it affects their ability to play, but most people find this unnoticeable or tolerable. We can calculate our approximate desired buffer size with this formula:
(Sample per second / 1000) * Desired Latency
So, if we are recording at 44,100 Hz and we are aiming for 20ms latency:   44100 / 1000 * 20 = 882 samples. Most audio devices do not allow arbitrary buffer sizes but offer an array of choices, so we would select the closest option. The device I'm using right now offers 512 and 1024 samples as the closest available buffer sizes, so I would select 512 first and see how this performs. If my session has a lot of tracks and/or several effects, I might need to bump this up to 1024 if I experience dropouts.
Now that we hopefully have a pretty firm understanding of what constitutes latency and under what circumstances it is undesirable, let's take a look at how we can reduce it for our needs. You may find that you continue to experience dropouts at a buffer size of 1024 but that raising it to larger options introduces too much latency for your needs. So we need to determine what we can do to reduce our overhead in order to have quality playback and recording at this buffer size.
Effects: A common cause of playback latency is the use of effects. As your audio stream passes through an effect, it takes time for the computer to perform the calculations to modify that signal. Each effect in a chain introduces its own amount of latency before the chunk of audio even reaches the point where the audio application passes it to the audio device and starts to fill up the buffer. Audition and other DAWs attempt to address this through "latency compensation" routines which introduce a bit more latency when you first press play as they process several seconds of audio ahead of time before beginning to stream those chunks to the audio driver. In some cases, however, the effects may be so intensive that the CPU simply isn't processing the math fast enough.   With Audition, you can "freeze" or pre-render these tracks by clicking the small lightning bolt button visible in the Effects Rack with that track selected. This performs a background render of that track, which automatically updates if you make any changes to the track or effect parameters, so that instead of calculating all those changes on-the-fly, it simply needs to stream back a plain old audio file which requires much fewer system resources. You may also choose to disable certain effects, or temporarily replace them with alternatives which may not sound exactly like what you want for your final mix, but which adequately simulate the desired effect for the purpose of recording.   (You might replace the CPU-intensive Full Reverb effect with the lightweight Studio Reverb effect, for example. Full Reverb effect is mathematically far more accurate and realistic, but Studio Reverb can provide that quick "body" you might want when monitoring vocals, for example.) You can also just disable the effects for a track or clip while recording, and turn them on later.
Device and Driver Options: Different devices may have wildly different performance at the same buffer size and with the same session. Audio devices designed primarily for gaming are less likely to perform well at low buffer sizes as those designed for music production, for example. Even if the hardware performs the same, the driver mode may be a source of latency. ASIO is almost always faster than MME, though many device manufacturers do not supply an ASIO driver. The use of third-party, device-agnostic drivers, such as ASIO4ALL (www.asio4all.com) allow you to wrap an MME-only device inside a faux-ASIO shell. The audio application believes it's speaking to an ASIO driver, and ASIO4ALL has been streamlined to work more quickly with the MME device, or even to allow you to use different inputs and outputs on separate devices which ASIO would otherwise prevent.
We also now see more USB microphone devices which are input-only audio devices that generally use a generic Windows driver and, with a few exceptions, rarely offer native ASIO support. USB microphones generally require a higher buffer size as they are primarily designed for recording in cases where monitoring is unimportant. When attempting to record via a USB microphone and monitor via a separate audio device, you're more likely to run into issues where the two devices are not synchronized or drift apart after some time. (The ugly secret of many device manufacturers is that they rarely operate at EXACTLY the sample rate specified. The difference between 44,100 and 44,118 Hz is negligible when listening to audio, but when trying to precisely synchronize to a track recorded AT 44,100, the difference adds up over time and what sounded in sync for the first minute will be wildly off-beat several minutes later.) You are almost always going to have better sync and performance with a standard microphone connected to the same device you're using for playback, and for serious recording, this is the best practice. If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device. Attempt to match the buffer sizes and sample rates as closely as possible, and consider using a higher buffer size and correcting the latency post-recording. (One method of doing this is to have a click or clap at the beginning of your session and make sure this is recorded by your USB microphone. After you finish your recording, you can visually line up the click in the recorded track with the click in the original track by moving your clip backwards in the timeline. This is not the most efficient method, but this alignment is the reason you see the clapboards in behind-the-scenes filmmaking footage.)
Other Hardware: Other hardware in your computer plays a role in the ability to feed or store audio data quickly. CPUs are so fast, and with multiple cores, capable of spreading the load so often the bottleneck for good performance - especially at high sample rates - tends to be your hard drive or storage media. It is highly recommended that you configure your temporary files location, and session/recording location, to a physical drive that is NOT the same as you have your operating system installed. Audition and other DAWs have absolutely no control over what Windows or OS X may decide to do at any given time and if your antivirus software or system file indexer decides it's time to start churning away at your hard drive at the same time that you're recording your magnum opus, you raise the likelihood of losing some of that performance. (In fact, it's a good idea to disable all non-essential applications and internet connections while recording to reduce the likelihood of external interference.) If you're going to be recording multiple tracks at once, it's a good idea to purchase the fastest hard drive your budget allows. Most cheap drives spin around 5400 rpm, which is fine for general use cases but does not allow for the fast read, write, and seek operations the drive needs to do when recording and playing back from multiple files simultaneously. 7200 RPM drives perform much better, and even faster options are available. While fragmentation is less of a problem on OS X systems, you'll want to frequently defragment your drive on Windows frequently - this process realigns all the blocks of your files so they're grouped together. As you write and delete files, pieces of each tend to get placed in the first location that has room. This ends up creating lots of gaps or splitting files up all over the disk. The act of reading or writing to these spread out areas cause the operation to take significantly longer than it needs to and can contribute to glitches in playback or loss of data when recording.

There is one point in the above that needed a little clarification, relating to USB mics:
_durin_ wrote:
If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device.
If you are going to spend that much, then you'd be better off putting a little more money into an external device with a proper mic pre, and a little less money by not bothering with a USB mic at all, and just getting a 'normal' condensor mic. It's true to say that over the years, the USB mic class of recording device has caused more trouble than any other, regardless.
You should also be aware that if you find a USB mic offering ASIO support, then unless it's got a headphone socket on it as well then you aren't going to be able to monitor what you record if you use it in its native ASIO mode. This is because your computer can only cope with one ASIO device in the system - that's all the spec allows. What you can do with most ASIO hardware though is share multiple streams (if the device has multiple inputs and outputs) between different software.
Seriously, USB mics are more trouble than they're worth.

Newbie Q. Re: Exporting audio in L8P with TDM...

Hey Folks,
1st time posting- sorry for making someone answer this again if it's been posted before... but I couldn't find the answer in a search, in the manual, or on any recent threads here.
OK, here goes:
Doing pre-production work in Logic 8 prior to exporting into Pro Tools for tracking/mixing/etc...
I'm using the DAE & DTDM engines, and it's been working great, although, I have to admit, signal flow in L8 has taken me a little bit to catch on to.
Went to export the tracks (12 loop/VI tracks [DTDM] and 3 mono audio tracks [DAE] ) and it will only allow me to export the 12 DTDM tracks "created" in Logic??? Can't seem to get it to export the 3 real tracks (recorded with mic/pre/etc...) no matter what I do...
The 12 VI tracks outputs are assigned to 2 stereo pairs (ESB 1-2 & ESB 3-4), which then feed 2 stereo Auxes (input=ESB 1-2/3-4, output= Out 1-2) and the 3 "real" DAE tracks are assigned directly to Out 1-2.
I've tried re-assigning busses, directly exporting the regions as audio, disabling inputs....NADA !
I'm sure this is probably REALLY simple, but I'm stumped!
Any help would be MUCH appreciated.
Thanks!
Rick

Still looking for an answer on this one....
Maybe the Q's worded poorly?
Basically I'm wanting to know how to export L8 DAE tracks for use in other apps. (Pro Tools in my case).
I've got a workaround (create DTDM audio tracks and physically move the audio files onto them, then export), but it doesn't allow for printing of the TDM plugs that were inserted on the tracks (which were a significant part of the sound of the track).
Any help is much appreciated!
I'll try back at the DUC as well...
Rick

Can airport express be used for all audio from my ibook?

I want to be able to route all of my computer audio to the airport express station conected to my stereo. This would come in handy when I want to watch DVD's on my laptop in my dorm room.
Airport Express Mac OS X (10.4.4)

Yes, you can with with Airfoil.
However, for watching DVDs, you'll want to use VLC in concert with Airfoil so that you can compensate for the audio delay that will occur otherwise. Open Airfoil and select VLC as the app whose audio should go to the Express. Click on the little button next to the speaker name and VLC will automatically open.
Then open the VLC preferences, select Audio in the list, and click on the Advanced checkbox at the bottom of the preferences window. Enter a negative number of milliseconds in the "Audio desynchronization compensation" field to advance the audio ahead of the video, which will offset the audio delay. Start with -3000 (3 seconds) and you'll probably have to tweak it from there. Then open the DVD in VLC and start it playing.
Yes, this is a pain in the butt, but it's the nature of streaming audio: the audio stream must be prebuffered for a few seconds to keep variations in the data transfer rate from interrupting the audio.

Premiere elements 12 - no audio

Longtime Photoshop Elements and Premiere user... now on v12. Premiere seems to work fine until I just realized there is no audio...
I can't hear audio in the preview playback
there is no audio track at the bottom of the screen
when I write out MP4 files, no audio is present.
Any ideas? O/S is Windows 8.1

Hi A.T.
Here are responses to you questions...
Did you update it yet from 12 to 12.1 via the opened project's Help Menu/Update? YES
You say that you are a longtime Photoshop Elements and Premiere Elements user. What was the most recent version of these programs that you worked with and on what computer operating system immediately before going to version 12 on Windows 8.1 64 bit? I've used 3 or 4 prior releases. I just upgraded from v9 => v12. v9 was running on Win7; v12.1 is running on Win8.1 (64-bit)
Are you working in the Expert or Quick workspace of Premiere Elements 12? Primarily Quick, but I use Expert when I need to manipulate sound
Because of the longtime in your description of your programs usage, I hesitate to ask, but I will anyway so as not to take anything for granted.
a. Are you working Run As Administrator as well as from User Account with Administrative Privileges? I am the only user of the machine, so my user account and Admin are one in the same.
b. Do you have the latest verison of QuickTime installed on your computer? Yes
c. What are you or the project setting as the project preset to match the properties of the source media?
d. What are the properties of the source media?
e. What are the details of your export to a .mp4 file?
Have you gone through the drill of checking the ASIO settings in preferences? Yes Can we assume that the no audio is only in Premiere Elements 12 on Windows 8.1 64 bit? Correct
Are the Volume settings (Adjust Tab/Adjustments Palette/Volume Panel) as well as audio clip rubberband) OK? Not sure...
Could you post a screenshot of your workspace where you say that there is no audio track? The audio track is physically present in the input sample, is just not audible. When I write out a MP4, there is no audio track present.
Have you attempted to delete the Adobe Premiere Elements Prefs file and/or the 12.0 Folder in which it exists? No... Nof familiar with doing that.

Audio connections to receiver?

Can I keep the AppleTV hdmi connection directly to the TV while using red/white audio cables to connect to my receiver for surround sound? It's an older receiver--not sure it has hdmi connectivity. Would this type of connection cause any problems?

jg051275 wrote:
I have been using a similar setup with my Atv, but I notice that the sound echoes when playing through both the television and the receiver... like the are not in sync. So I usually just mute the tv when I have the receiver on.
Exactly what you should be doing if playing back audio via a receiver.
Anyone else notice this or know how to remedy the issue with an older receiver that has composite only?
Does your receiver offer any output delay options if it has any menus?
Modern TVs to a lot of video processing which can lead to a lag of 10's of msec between the input signal and the picture/audio output by the TV.
If the processing causes a long enough lag the receiver can be outputting audio ahead of the TV - we all have different tolerance to such lip sync issues but the longer the image processing lag the more likelt you are to notice the audio/video are out of sync between receiver and TV. Short delays may not be noticeable at all.
Many receivers allow a delay to be applied to counter this either to all or individual surround channels.
My older Yamaha unit doesn't have an overall delay function but I rarely notice issues.

Ducking doesn't work on added audio track

I have an iMovie 09 video clip of people dancing to music in the background. I attached a separate audio clip of music to the video (it appears in green beneath the video).
At a certain point in the video, I want to fade the added music clip so you can hear what's going on in the video. So I went to "Audio Adjustments" on the video clip and checked the Ducking box to that the music on the added audio clip would fade at that certain point.
But it doesn't fade at all. How do I fix this? Thanks.

You should be able to trim expanded audio in previous versions, also. Then you do trim, the audio will only turn dark, not get physically shorter, when you Expand Audio Components. When you Expand Audio, it will physically change the length of the audio section. One thing to remember is that when you Expand Audio/Video, the whole clip is selected by default, click on the head or tail of the audio first. If it is red, you can not expand it out, but can trim it in.
Tom is correct about ROLL edits, they only work in FCPX 10.1 and later. Only works with clips in a Storyline. Only work with audio that has enough Handles to accoedate the Roll edit.

Audio-video out of sync on iPod Touch 5G

My iPod Touch 5G audio to video is out of sync.

1) Video's inside Facebook can have the audio lead the video by 15 secs.
2) Video live Apple product event feeds can start with 2 1/2 mins with audio ahead of video and gets more spread out as time goes on. As much as 5 minutes apart towards end of live feed event.

NTSC DVD audio out of sync

Hi there...
I have just compiled and built an NTSC DVD using Encore 2. The original videos were PAL and I converted them to NTSC using Canopus Procoder 2. They were uncompressed and used PCM 16bit audio at 48khz.
The compiling in Encore was fine, everything previewed perfectly (still does). The burned disc however has all of the audio 'ahead' of the video by about half a sec. Alot of the vid is high motion and transitions are set to the musics beat alot, so you can imagine how annoyed I am. I am reviewing the disc on my laptop, the same one I use to preview all of the discs in the past which have been fine.
The PAL version that I made is fine. I have made NTSC copies before and not had a problem. There are no gaps in the timeline so I am a little stumped at why it is not working.
Any quick help would be much appreciated.
Simon

Simon.
More details please.
Project Structure
Layout, Audio Format in final disc, etc etc.

I have a serious issue with static (clicking sounds) when I use Netflix streaming along with any movies I rent via AppleTV, and yes I use a optical audio cord. Any suggestions?

Was is most interesting, is that if I watch other content like MLB TV using the sound bar and optical cable, I don't have this issue, it's all very strange. Any suggestions?

Perhaps it's obvious to you, but what would I adjust in the midi settings? There's nothing obvious that speaks to this out of sync issue.
FYI, I just plugged my JBL creatures into the audio out (same physical jack as the digital out, though of course they use different technologies) and there is no sync problem with the audio played through them. Clearly it's an issue with the digital out, just wish I could figure out if it's my stereo or something in the computer. I have no other devices that accept a digital input, so can't swap the stereo out for anything else.

How to use my external processor effects?

I have a lexicon processor and I want to know how to use it with logic.
Thanks
g5 ma' os Mac OS X (10.4.4)

I hope this info can help you.
The main Problem is that Logic don't have an delay compensation plugin like in nuendo or cubase "external FX"
Here is the info to compensate the delay coused by the D/A and A/D converters:
6.30 How do I set-up record / playback / monitoring delay?
Subject: Record delay vs send/return using external effect units
You'd think that with modern multichannel audio interfaces and a modern, professional audio sequencer like Logic, this would be a piece of cake, right? It turns out that there are a number of potential traps, all to do with Logic's highly inadequate record delay compensation. What follows is a run through of the general setup procedure, using my RME Multiface.
Basic Record/Playback delay Setup
Set up a 4/4 audio click track (trim the sample starts so they are right on the beats). Use real audio, not a software instrument. Rerecord the main outs to a new track via a hardware loopback cable (i.e. cable your audiocard's outputs into its inputs). Measure the clicks with reference to bars/beats in the Sample Editor - the clicks should be (but may not be) recorded right on the beat.
If the audio driver has a record delay parameter in samples, use that to adjust. If not - use the ASIO Buffer delay "IN" for coarse adjustment (multiples of buffer). Leave the ASIO Buffer delay "OUT" at zero. Use the main driver playback delay for final fine adjustment (samples). Do NOT use the Arrange window's delay parameter - it's in ticks and thus tempo-dependent!
So if you are playing in time with a prerecorded track, your playing is recorded in the correct position to preserve your exact timing on playback.
The use of the playback delay to compensate is not ideal, as it will mess with the playback timing vs displayed position of any audio - this has an adverse effect on fine editing, and MIDI-to-audio sync - but Logic unfortunately does not provide a sample-accurate record delay adjustment, and has not done so since version 3.5...
Effect Return/Record Setup
You definitely want to avoid monitoring external FX returns through Logic if at all possible, since that would add 2x the audio buffer worth of latency to the return, which will adversely mess with the sound of any time-based effects (i.e. just about everything). So, monitor your external FX returns at source, or through direct hardware monitoring - in my case that's through RME TotalMix routed to the main outs for (near) zero latency.
You probably want to be able to record the external FX outputs into Logic (to free up the FX for other uses) and have it play back exactly as it was monitored, right?
Using your 4/4 audio click track, panned left, send to an external delay (approx 1/2 beat, single repeat). Pan the delay return R at source and monitor as above. Rerecord the main outs to new track via a hardware loopback cable. Measure the number of samples from a click to its delay in the Sample Editor
Now record the delay return signal to a new track. Playback just the audio click panned left & the recorded delay panned right. Rerecord the main outs to new track via hardware loopback cable. Measure the number of samples from click to delay in Sample Edit.
Both measurements must be the same for accurate recording of FX returns, but it's likely they won't be, probably because the playback delay has been messed with. Compensate by inserting a sample delay on the FX return Input Objects in Logic. Since you're not monitoring the FX returns through Logic anyway, the input delay will be recorded but not monitored (in 5.2+, but if you use an earlier version you're hosed since effects are not recorded).
Record the delay return again, play it back with the click, rerecord the outputs and measure again. Adjust the input sample delay until the recorded-&-played-back delay position is identical to the monitored delay.
Live Input Monitoring/Recording
For accurate timing, monitor any live inputs at source or through direct hardware monitoring (TotalMix), the same as for FX returns. When recorded, playback timing will be accurate. Sends from live inputs to external FX can also be applied in at source or in TotalMix. No problem.
Live Input Monitoring Through Logic
Here's where the problem starts. If you also plan to monitor some live inputs through Logic - to add Logic FX, or to control & automate the live inputs, or to add live inputs to a bounce, etc - then you'll be monitoring the live inputs with a latency of 2x the audio buffer size (or more if applied processing induces further delays). Therefore when you record a live input, on playback it will be early by that amount, since the record/playback delay is set up to compensate for zero-latency monitoring. What you heard live is not what you get on playback.
There's no set-&-forget way around this, since Logic won't let you apply a sample delay to an input without the delay also being applied to the input's monitor output from Logic. So using the same input delay trick you applied to recording the FX returns won't work - you'll wind up monitoring with even more delay, which will need yet more input compensation, and so on. You can't use the record/playback delay to compensate, because that would screw up the recording of source-monitored material.
You could conceivably monitor everything through Logic at all times, and use a single record/playback delay to compensate for all of it (with the editing & MIDI-to-audio sync shortcomings discussed, but on a larger scale as considerably more compensation is required), but that will screw with the sound of any time-based external FX as noted above.
So assuming you stick with source-monitoring the external FX returns, there are 2 options for correct playback timing of any recorded tracks that were monitored through Logic while recording:
1 - Output all such tracks to a bus, and place a Sample delay on that bus to compensate. This will correct the playback you hear, but it won't correct the bad positioning of the audio.
2 - Physically move the recorded audio later to compensate. Uh-oh - Logic's Arrange window is not sample-accurate. And it's not possible to move a newly recorded audio region later in the Sample Editor without adding samples to the start of the file, which is a tedious process if you've just recorded a number of tracks. You'll just have to get it as close as you can in the Arrange - bear in mind that ticks are tempo dependent, so you have to calculate the number of ticks based on the current song tempo (let's not even begin to discuss Audio vs tempo changes in Logic), or use the smallest SMPTE nudge available. No fun at all.
If anyone has any other suggestions, I'm all ears...
So what about OS-X?
In Logic under OS X, the CoreAudio driver setup panel doesn't have any record/playback delay setup. If you're optimistic, you might interpret that as an indication that it's all done automatically by CoreAudio and the driver. But given Emagic's history in this area, what's the bet it's currently a big fat inaccurate mess? Rumblings from the Mobile i/o list seem to indicate this...
G5 dual 2,5 Mac OS X (10.4.4)

Can N200 battery be used for N100 laptop

Hi,
One of Lenovo resellers suggested that N200 battery can be used for N100 laptop as well. However, N200 one is 11V and N100 one is 10.8V. Otherwise it fits in the laptop design.
Any recommendations or suggestions from experts?
Thanks
Ahmer

Yes, you can with with Airfoil.
However, for watching DVDs, you'll want to use VLC in concert with Airfoil so that you can compensate for the audio delay that will occur otherwise. Open Airfoil and select VLC as the app whose audio should go to the Express. Click on the little button next to the speaker name and VLC will automatically open.
Then open the VLC preferences, select Audio in the list, and click on the Advanced checkbox at the bottom of the preferences window. Enter a negative number of milliseconds in the "Audio desynchronization compensation" field to advance the audio ahead of the video, which will offset the audio delay. Start with -3000 (3 seconds) and you'll probably have to tweak it from there. Then open the DVD in VLC and start it playing.
Yes, this is a pain in the butt, but it's the nature of streaming audio: the audio stream must be prebuffered for a few seconds to keep variations in the data transfer rate from interrupting the audio.

Recorded audio ahead of physical recording

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