Couple of ideas. 1) sometimes tempo fluxuations can seem abrupt if there are not enough beat mapping markers. See if you can identify more sub beats to create a more gradual accelerando. 2) Assuming this is not a classical piece, there are probably sections of the song that repeat. Can you replace the part where the tempo begins to speed up with a similar section from earlier (or later) in the song. Sometimes creative editing does the trick. 3a) if you need to use the audio quantize, try quantizing only the sections you want versus the entire track. 3b) Sometimes the audio engine being used makes a huge difference. Acoustic guitar is one of those things where its very difficult to do without artifacts. I have alternate algorithms that I select, Speed from SoundToys can sound good. Also Izotope makes Radius. bd

I'm setting up multiple screensets per the tutorial, but the second screenset is not automatically moving to the selected region per the behavior in the movie. Is there a Setting I'm missing? It seems the focus of the zoomed in screenset is always wherever the playhead is, it remains oblivious to anything I select in the above screenset. thanks! bd

Ah, Thanks Steve. bd

Like many of the tutorials it's great inspiration for coming up with crazy new ways to manipulate audio. I made a small variation on the Panner circuit in this tutorial and thought I'd share. First, Create the circuit as it is at the tutorials end. Next, Insert an extra transformer between the Panner Instrument and the Arpegiator. Set the condition status to Note and the operation velocity to Sub 42. Now route cables out the Panner instrument, the Arpegiator, and the first transformer (the condition splitter, not the one we just made) all to the instrument. The effect is that the First note or chord played will be louder than the arpegiated panning aftermath. It's a nice way to keep the the focus on the played part but still get that panning goodness on help notes. it's a little more ambient in effect and less obvious. bd

Waveburner does not embed any codes or text into the audio files. The ISRC code, UPC Code CD Text etc. is purely for the CD Burning process. There are applications that can embed additional text information into the headers of the audio files. For example, Sample Manager from Audiofile-Engineering can embed an entire customizable list of "properties" into the sound file. It is important to note though that most applications will not be able to read or even see the additional text data. Sample Manager and Sound Miner would, And in the case of a few of the properties, Like Album and Artist for example, iTunes will see the data. But Waveburner will not. It may also be worth mentionioning here that the CD text you embed will not be seen by iTunes without the help of a 3rd party plug-in that specifically imports CD Text into iTunes prior to importing the Disc. I use this. http://dougscripts.com/itunes/scripts/ss.php?sp=cdtexttocdinfo hope that helps.

First. There is no industry standard app that I'm aware of. Sound Miner is about the closest I can think of, and it's not really being used to watermark. Header data is easily stripped out of the file so it's only good in terms of an in-house tracking method. Basically there is no audio watermark (AFAIK). ISRC codes are a way to identify specific songs. That's a standard in that many digital distribution systems require the song have an ISRC code. What that does? I don't know. Second, no. The whole thing is much more complicated than that. If you have music getting placed in commercials and movies, the best thing to do is have someone who's job is to track and submit it. a lot of this still comes down to people dealing with people. There's no machine tracking everything and doling out appropriate $$. bd

Having the Example of the how the internal mixer works in regards to 32 bit processing and clipping was entirely worth the price of admission. I've been running some of my own tests with various plug-ins parially because I can't really beleive it, but it's true and amazing. I cranked up the input and output of Ozone 4 for almost 40db of gain, popped a Gain plug-in after it and got a pure sine wave. I wish I had understood this years ago. I think because I came up through the final years of primarily analog studios I had just been in the habit of seeing clipping as clipping. You clip a channel or piece of gear in the analog domain, and you get distortion, sometimes good, sometims not, but there's nowhere to bring the gain back on the other side to remove it. So thanks for the mental can opener! bd

I also just couldn't resist running some of the same tests in PT HD. The results were Intriguing to say the least and I think goes a long way into explaining why I like the sound of analog summing when I mix on that system. Whereas when I mix in Logic I don't really hear the need for the summing setup. I've always understood that PT HD is 48bit fixed and read and listened to arguments regarding the whole 32bit float versus 48bit fixed thing, but to be able to demonstrate in a quasi scientific method to myself what it comes down to functionally is pretty darn empowering. I've never given the fact that PT LE operates with different math than PT HD much thought. But I think it changes things dramatically for me. I've been mixing (and until about a year ago mastering) in TDM since TDM was created, and clipping has always been a bad thing, I now understand why AND can demonstrate it. As a caveat there are many other reasons why an HD system has its strengths... This is just all really interesting. bd

Pro Tools does not record 32 bit files. If the files are from Pro Tools, then they will be 24 bit or 16 bit. Logic will import those without any conversion just fine, I do that all the time. You will only need to convert the sample rate if that doesn't match. Logic will display what the files are (sample and bit rate) in the import dialogue. if the files are 32 bit, the process gets a little more complicated. (I've had bad luck anytime I've tried doing anything with 32 bit files) You would need to find a way to convert them to 24 bit before Logic would be able to see them for import. There are a few options that might work for that. Cubase, Peak and Sample Manager would be the first that come to mind. Hinder Quality? That depends on what and in what context. The simple act of converting isn't inherently bad. The only way to know for sure is trust your ears. Do you hear a difference after converting? Is it acceptable? bd

I have problems using the copy / paste audio config in the mixer window. Mainly, if the session you are pasting into isn't the exact same layout, then the paste can really messes things up. Mostly I wind up using the save channel strip settings and just manually loading appropriate channesl. A pain to be sure, but it works. I hope for the snow leopard update they either bring the config window back, or provide a more detailed importing process for mix engineers.