Matt Barber wrote:
Hey Greg,
I wonder what would happen if you split your 8-channel files and played them simultaneously as 4-channel soundfiles...
Yeah, this is how the piece was originally created -- the idea being, if all "mixing" happened in real time, it would be easier to effect musical changes during rehearsal.
or, if it's always running on your computer, you might try storing half of your soundfiles on one disk and the other half on another -- it's hard to know whether the disk is going bonkers or readsf~ breaks.
Good thinking. The soloist is considering an upgrade to dual 10,000rpm SCSI drives -- a wise move given that mine is not be the only work in his repertoire requiring this level of computer performance.
Depending on the size of your files you might bite the bullet and load them into tables...
I tried this, but it only works in a few cases since the files are generally too large for tables.
At any rate, you should have three files -- one is a generic abstraction that shows the general method. The other abstraction (playback_8ch_fade) is based on your first patch, and is generalized to play any 8-channel soundfile -- hopefully the comments in them are useful.
This is *extremely* helpful. My previous approach involved the creation of unique abstractions for slightly modified instances. Your "all in one" is far more flexible and better suited to real world use.
I kept your 6<n<56 ms "humanization" in the file, as well as the throws to ch1-ch8 (I'd probably use outlet~ more often than not, but this is fine if you know how you need to set up the rest of the patch).
Yup, you hit the "humanization" nail on the head. In certain sections of the piece, the computer builds complete musical gestures in real time. Depending on where it is in the score, the patch chooses the appropriate soundfile type and selects a specific file to trigger from a predetermined list wherein all files are similar, but never identical. (For obvious reasons, subtle volume and timbre changes are important when attempting to create an organic and richly varied performance environment). Randomizing start times between 6 and 56 ms seems to provide a natural "ensemble" feel in these instances.
It uses the same general method as the generic patch, but adds some other goodies I would feel obligated to provide if I were giving the abstraction to someone to use... but maybe it's way overkill for personal use, or inside a patch where nobody's gonna see it. It has some basic type-checking and conversion, but no error printing, which I would normally do if I had the time or were building a library. It also has a small example of some dynamic patching, which might better be left out, and could maybe even be avoided in this example (nothing comes to mind instantly)... I like having the option to change things on the fly, though, so I use this kind of thing in my own patches all the time. Let me know if it's even readable. The third file (marked "revised") is an example of how to use the bigger abstraction. I haven't fully debugged it all, and lots of optimizations could be made all over the place but I think it should work as an example patch.
Fantastic. A wonderful abstraction tutorial. I'll be sure to post my "final solution" to the list once I find what works best.
Of course others are welcome to comment if it sucks, or use any of it if they find it compelling. =o) Let me know if this helps out, but I've a feeling your problem is deeper than any method for using [readsf~].
Agreed. And I'm surprised there aren't others running into this problem with 8-channel 88.2/24 interactive patches like this.
Of course, the 8-channel environment is useful for its ambisonic and other spatialization potential, and one solution that works well (for certain musical situations) is to spatialize monophonic soundfiles in real time. This is a great solution for reducing performance demand on the hard drive, but quickly becomes expensive in CPU cycles...
Best, G
On Mon, Jul 14, 2008 at 12:51 PM, Dr. Greg Wilder gregwilder@orpheusmediaresearch.com wrote:
Matt Barber wrote:
Date: Sat, 12 Jul 2008 13:56:54 +0200 From: Damian Stewart damian@frey.co.nz Subject: [PD] how to avoid (most/many/some) readsf~ dropouts To: PD-List pd-list@iem.at Message-ID: 48789C06.4030401@frey.co.nz Content-Type: text/plain; charset=ISO-8859-1; format=flowed
hey,
one thing i've noticed with readsf~ using it in my own live-performance sets is that doing this
[symbol blahblahblah.aif] | [open $1, bang( | [readsf~]
sometimes causes dropouts. but if you go
[symbol blahblahblah.aif] | [t b a] | ____ |
[del 50] [open $1( | __________/ |/ [readsf~]then you remove (all/many/some) of the dropouts. i haven't extensively tested this, but anecdotal evidence seems to suggest that it works.
I think this is because the former method may not let the buffer fill before you start playing. I usually don't like the second method, either, because of the 50 ms delay (which isn't much, but I'm finicky about that kind of thing). Depending on the setup, I usually prefer something like the following:
[loadbang] | | [r open] | / [open file.ext ([0 ( [1 ( | | __ | [t b f ]_____\ | | [readsf~ ] [s open] | [s open]
I hope things line up okay there... at any rate, you load the buffer at the beginning, or if your piece/performance has a master clock, a couple of seconds before you need it. Then for rehearsal, if you need to stop the file you can use a trigger to reload it immediately. Also, if you need to play the file again, you can send the "open" message when the file is done playing. This makes the whole process a little more "front-loaded," so that the soundfile is always "open" no matter what. My intuition is that it's more robust than other schemes I've tried. I haven't tested it very hard, though, as I've only ever needed up to 9-10 simultaneous 96k files... maybe it would be neat to find some old, slow hardware to see how far different methods can be pushed. =o)
More advanced for stopping would be something that fades out the sound with a [line~] over 20 ms or so, and then sends the stop message to [readsf~] after a comparable delay -- that way you won't get an annoying transient upon stop -- same deal with a fade in if you are starting in the middle of the soundfile, but without the delay. I usually wrap the whole thing in an abstraction -- I keep three or four different ones lying around with different properties for different situations, and I'm happy to share with anyone who might find them useful.
Thanks,
Matt
Matt and Damian,
In working on a recent piece for marimba and 8-channel computer (24bit/88.2kHz), I consistently experienced intermittent clicks and audio dropouts -- even on high-end hardware running GNU/Linux. Increasing the readsf buffer and the time between file load (open) and playback (readsf) helped some, but not enough. Since the premiere last month, I've been rebuilding the abstractions for better efficiency, but am still not happy with the results -- and before the soloist can safely tour the piece, I need to work out a more robust solution.
I've attached the latest version of my "basic" playback (w/fade) patch for suggestions/comments... Unfortunately, your ascii patch didn't line up, would you mind posting an example patch that shows your method?
Best, G
-- http://www.orpheusmediaresearch.com/ http://www.gregwilder.com/ +1 215-764-6057 (office) +1 215-205-2893 (cell)
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