Jean-christophe Boulay
Forum Replies Created
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ARDagent has always given that message, as it should. Remote Desktop has its own set of permissions which are not alterable by DU. You probably never noticed because it was hidden among all the other repair messages. It does not prevent further permissions repair in any way. That doesn’t solve your problem, though.
You don’t mention DiskWarrior in your post, which is probably the most important tool to keeping a sleek system. DW should be run whenever the system acts up a bit, or on any maintenance schedule you run. It corrects errors in the databases that DU can’t. It also has S.M.A.R.T. and permissions repair utilities so you can do all your disk checking in one place. DW must be run from another boot drive to work but comes with its own bootable DVD.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
I’ve seen these bands on many h.264 exports. The funky thing is you don’t really see them if you’re squarely in front of your LCD, they only appear if you look off-axis, which explains why you see them so readily on your laptop. The image you posted looks fine on my monitor, but if I turn it 45 degrees off axis the bands appear.
I’ve personally only seen this issue with h.264, in either an .m4v or .mov container, when exporting in a resolution smaller than the source through compressor. And not on all of them. I admit I never really looked too hard for a solution, as we use that codec mainly for client approvals. The clients that did see the bands were comforted with an export in another codec or a full-res screen grab, where they disappeared.
So I don’t have a solution for you, but it’s not a hardware issue with your laptop.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Amy,
The setup you’re describing is not optimal. The output you should be using on the MacPro is the Line Out, which is at the rear. It’s the one with a little speaker symbol next to it. That output should give you more level, which should help.
The other consideration is the cable’s length. The type of cable you’re using is correct, but such a cable is unbalanced. Long story short, if your cable is too long (over 15 ft) and/or is in an environment with a lot of electromagnetic interference, the cable will pick up interference and transmit it down the line with your sound. With the connection you’re trying to make, the only real solution is keeping the cable length as short as your needs permit. Try changing outputs before futzing around with the cable, though, as that could be the ticket. You may also want to try plugging another source that you know works into that input on the mixer to eliminate that as a cause of your problems.
If it doesn’t work, report back and we’ll go to level 2.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Good point on third-party interfaces, Ty. There are some interfaces that don’t play as nice with PT9, where you’ll have to go to the interface’s control pannel to change some settings that should be changed from PT itself. I’ve had a friend complain about his Tascam interface not changing bit depth and sample rate with the PT session. I recall a page in Avid’s support section that listed problematic devices but can’t seem to find it now. You can go drop into the PT9 forum (duc.avid.com) and list your planned setup to see if others have been running it successfully. They won’t bite.
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
If you’re going to be interfacing with MC5, Pro Tools is the absolute no-brainer. While other DAWs have a solid foothold on the music production side, in post PT is king. Of course, some people use other software but you’ll never see studios exchanging Logic sessions. It’s PT sessions or OMF/AAF.
PT9 should play nice with MC5 on the same computer. There was a time when maintaining a healthy PT rig was a tough job, but frankly, now it’s not. It’s a lot less finicky than it once was. If you’ve ever worked with hardware mixers, PT’s audio routing scheme makes sense instantly as well.
So if your question was “What audio software is good?” the answer could’ve been anything. If you’d asked about favorite software for music production, my answer would be completely different. Since you ask about standards in post, it can only be PT. Standard plugins are harder to nail down.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Steven,
Audio does not have frames per second. It has samples per second, in most cases concerning audio for video, either 44100 per second or 48000 per second, a.k.a. 44.1kHz and 48kHz.
The first major error people do with audio is not checking their sample rate and importing one into a project of the other or mixing them in the same project. Ensure your sequence audio settings and your audio files specs match. And that you’re working with uncompressed audio, of course.
The other main source of this kind of issue in FCP is when you’ve never changed the default setting (that window that pops up the first time you open with “DV NTSC 48kHz” selected) . If you’ve never changed that, FCP can apply a pull-up or pull-down to the audio for some reason, primarily because FCP hates audio, I think. Seriously, this issue demonstrates why you should never have a video frame rate associated with audio. It confuses people. Even the FCP coders got it wrong. I think simply having a correct Easy Setup set up solves this, but I’m not 100% on the fix. Citation Needed.
I’d wager your problem is one of the above.
IHTH
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Molly,
Really, the best you can do is get rid of some of the noise. A telephone has a frequency response of 300Hz to 3400Hz (in North America, at least), so most of the frequency ranges you’d be EQing to affect voice tone are not even present in the signal. If you boost above 4kHz, you’ll just be boosting line noise, as there was never any audio there to be boosted. I find it quite frustrating to work with phone audio.
What I normally do with such signals is try to eliminate as much noise as possible, first by EQing out the frequency ranges containing only noise, then by passing the result through a full-band noise reducer, like Z-Noise. I have to check myself not to go overboard at this point. Leaving a bit of noise in there smoothes the overall result a bit. The phone signal is already very compressed, so playing around with dynamics is not likely to get you anywhere. A bit of additional EQing will let you address the voice itself at this point, but the scope is very limited.
Basically, you’ll be scratching your head, wondering what else you could possibly do to make it better.
The fact is, very soon that answer is “nothing”.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
My technique still works perfectly. I don’t use any special software or services, just good ol’ Safari. This will always work for almost any video embedded in a web page.
-Open Safari and go to the page with the video you want.
-Open the Activity window in Safari. This will list all the elements loaded to display the page. If you’ve visited many pages, there will be a disclosure triangle for each of them. Open the one for the page you want to steal video from.
-Find the video among the other elements. It’s usually much much bigger than anything else. Everything will be in kB and you’ll probably have your video only displaying MB.
-Click on that element and command-C.
-Open your Downloads window in Safari and command-V your link in there.
-You’ll start downloading the file. For YouTube, the file will be called “videoplayback.flv”.
-Rename and convert.Safari is the best browser for this, as the activity reports are so easily accessible compared to other browsers that hide them in developer modes or some such nonsense.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
Hi Ty,
LKFS is a newer level measurement measure. It’s “Loudness, K-Weighed, relative to Full Scale”. The process takes all the channels of your multichannel audio signal, dumps the LFE, applies a filter to all channels (the K-Filter, which mostly cuts some low end and boosts about 2dB above 4kHz), measures the average power (Leq) and sums the channels back up, with some weighing. The surrounds are boosted 1.5dB before summing, for example. All this process is done within the metering tool as long as it conforms to the ITU-R BS.1770 standard. The measure is in decibels, but is reported as LKFS alone, not dB LKFS. I know TC Electronics have updated their metering tools with this algorythm. I like AudioLeak for its simplicity (https://www.channld.com/audioleak/).
I haven’t seen a spec sheet with LKFS on it yet, but some contacts at local broadcasters say that shouldn’t be too long, especially in the US where this will become the law by december. We’ll probably have to specify Dialog Level when delivering broadcast audio, so the DialNorm can be set accordingly by the operator. Either that or we’ll receive precise Dialog Level specs to respect. Because of the dumping of the LFE and the filtering and weighing, LKFS is really hard to guess based on more traditional measurements. You really have to measure it, which probably means new tools. And we all know how exciting it is to invest in metering! Clients go crazy for that stuff.
IHTH,
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com -
On this same subject, the ATSC has just published a new version of its CALM recommended practices. Most of the document is the same but there are additional annexes relating to techniques to be used on non-AC3-coded signals, so bypassing the standard LM100-DialNorm workflow. Basically, we fall back to a broadcaster’s LKFS norms in that situation.
Of course, it’s rather dry and techy but important reading for anyone who’ll be producing broadcast audio in the next few years. Pass the word.
https://www.atsc.org/cms/standards/a_85-2011a.pdf
JC Boulay
Technical Director
Audio Z
Montreal, Canada
http://www.audioz.com