Re: [PD] how to avoid (most/many/some) readsf~ dropouts
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 6n56 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 [EMAIL PROTECTED] wrote: Matt Barber wrote: Date: Sat, 12 Jul 2008 13:56:54 +0200 From: Damian Stewart [EMAIL PROTECTED] Subject: [PD] how to avoid (most/many/some) readsf~ dropouts To: PD-List pd-list@iem.at Message-ID: [EMAIL PROTECTED] 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
Re: [PD] how to avoid (most/many/some) readsf~ dropouts
Matt Barber wrote: 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... Right. You can split the difference, though, if you're using ambisonics, provided you're using B-format (wxyz). You can do all your ambisonic encoding and room simulation ahead of time, and then do the decoding in Pd. This way you'd only be reading four channels at a time, and the conversion from B-format to 8-channels is a fairly inexpensive set of multiplies and adds (a little more expensive if it's a cube rather than an octagon array, I think, since you could discard the Z harmonic with the octagon; in Pd a cube decode could be on the order of 24 +'s and as few as 4 *'s, most of the adds taking place in connections as the vectors are automatically added) -- you could easily make an abstraction to just put on the end right before you send it to [dac~], since b-format streams should mix linearly. I'm sure there are externals which could do this more efficiently than an abstraction (loathe as I am to use externals when there's an easy abstraction solution). In this setup, normalization becomes a little harder, though. It would also be useful if you later wanted to do some simple ambisonic panning of the solo marimba throughout the array - you'd have half the architecture you'd need for it, and B-format encoding of two or three streams is not gonna break the bank (unless you were doing some kind of full-on room simulation on top of it). The point is moot if you're using 2nd-order ambisonics, though, or if you've already spent a lot of time mixing and normalizing. Great points all around. Of course I spent a great deal of time considering a range of similar approaches before I began work on the project. The commission dictated a high-resolution, 8-channel cube array, and the decision to avoid B-format came down to the fact that I wasn't happy with reverberation quality produced by the available csound ambisonic and spatialization algorithms. I knew I was giving up a certain amount of flexibility by directly rendering the files (using csound and a custom java-based preprocessor), but it seems I didn't quite anticipate the heavy demand the files would put on the playback system. G ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] how to avoid (most/many/some) readsf~ dropouts
Matt Barber wrote: Date: Sat, 12 Jul 2008 13:56:54 +0200 From: Damian Stewart [EMAIL PROTECTED] Subject: [PD] how to avoid (most/many/some) readsf~ dropouts To: PD-List pd-list@iem.at Message-ID: [EMAIL PROTECTED] 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) #N canvas 65 224 902 519 10; #X msg 86 198 1; #X msg 129 198 0; #X obj 175 128 bng 15 250 50 0 empty empty empty 0 -6 0 8 -262144 -1 -1; #X obj 129 178 r stop; #X obj 86 105 + 6; #X floatatom 86 125 5 0 0 0 - - -; #X obj 175 373 *~; #X obj 383 147 loadbang; #X obj 441 147 r reset_vol; #X text 401 127 master volume; #X obj 86 84 random 50; #X obj 243 373 *~; #X obj 174 395 throw~ ch1; #X obj 242 395 throw~ ch2; #X obj 312 373 *~; #X obj 380 373 *~; #X obj 311 395 throw~ ch3; #X obj 379 395 throw~ ch4; #X obj 574 145 delay 500; #X msg 383 168 100; #X msg 441 168 100; #X obj 667 241 \$1; #X obj 395 351 vline~; #X obj 328 351 vline~; #X obj 258 351 vline~; #X obj 191 351 vline~; #X obj 86 143 delay; #X obj 448 373 *~; #X obj 516 373 *~; #X obj 584 373 *~; #X obj 574 224 dbtorms; #X floatatom 574 185 5 0 0 0 - - -; #X obj 574 203 * 1; #X obj 652 373 *~; #X obj 667 351 vline~; #X obj 600 351 vline~; #X obj 531 351 vline~; #X obj 464 351 vline~; #X text 599 204 (vol%); #X obj 447 395 throw~ ch5; #X obj 515 395 throw~ ch6; #X obj 583 395 throw~ ch7; #X obj 651 395 throw~ ch8; #X msg 175 151 open 8ch_acheron_blasts.wav; #X msg 574 165 0; #X msg 667 262 \$1 3500; #X obj 667 283 unpack f f; #X obj 724 304 + 1025; #X text 719 262 fade time in ms; #X floatatom 724 325 5 0 0 0 - - -; #X obj 441 189 delay; #X obj 441 209 bng 15 250 50 0 empty empty empty 0 -6 0 10 -262144 -1 -1; #X obj 175 67 r acheron_play; #X
Re: [PD] how to avoid (most/many/some) readsf~ dropouts
Damian Stewart wrote: Dr. Greg Wilder wrote: 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? first thing i notice when i open it up is this: on the left hand side, you've got a 'random 50' and a '+ 6', going into the delay. this means the minimum delay will be 6ms, which i doubt is long enough. why the random? why not just hardcode 50ms in there? or at least go 'random 30' and '+ 40' or something. hth, d Indeed. Sorry about that -- I pulled the example from a larger abstraction which added another 2000 to the delay amount, but the +2000 didn't get copied. At any rate, the randomized load time is there to prevent the patch triggering multiple files at exactly the same moment. My understanding is that PD's execution order doesn't allow simultaneous events to collide, however I have found I get better results when I leave a few extra milliseconds between simultaneous triggers. Perhaps it would be smarter to use a hard coded [t b b b]? Also, I should point out that the dropouts I experienced happened at different points during playback. Sometimes near the beginning, other times 50 seconds in... G ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] how to avoid (most/many/some) readsf~ dropouts
Damian Stewart wrote: Dr. Greg Wilder wrote: 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? first thing i notice when i open it up is this: on the left hand side, you've got a 'random 50' and a '+ 6', going into the delay. this means the minimum delay will be 6ms, which i doubt is long enough. why the random? why not just hardcode 50ms in there? or at least go 'random 30' and '+ 40' or something. hth, d Here's a cleaner version of the example patch -- without the random delay and a larger readsf~ buffer. Still no joy -- even with 92.9 ms latency. I'm beginning to think my system simply isn't fast enough to play more than 10 (or so) channels of 24bit/88.2kHz sound at a time... G -- http://www.orpheusmediaresearch.com/ http://www.gregwilder.com/ +1 215-764-6057 (office) +1 215-205-2893 (cell) #N canvas 311 207 902 479 10; #X msg 86 174 1; #X msg 129 174 0; #X obj 129 154 r stop; #X obj 175 349 *~; #X obj 383 123 loadbang; #X obj 441 123 r reset_vol; #X text 401 103 master volume; #X obj 243 349 *~; #X obj 174 371 throw~ ch1; #X obj 242 371 throw~ ch2; #X obj 312 349 *~; #X obj 380 349 *~; #X obj 311 371 throw~ ch3; #X obj 379 371 throw~ ch4; #X obj 574 121 delay 500; #X msg 383 144 100; #X msg 441 144 100; #X obj 667 217 \$1; #X obj 395 327 vline~; #X obj 328 327 vline~; #X obj 258 327 vline~; #X obj 191 327 vline~; #X obj 448 349 *~; #X obj 516 349 *~; #X obj 584 349 *~; #X obj 574 200 dbtorms; #X floatatom 574 161 5 0 0 0 - - -; #X obj 574 179 * 1; #X obj 652 349 *~; #X obj 667 327 vline~; #X obj 600 327 vline~; #X obj 531 327 vline~; #X obj 464 327 vline~; #X text 599 180 (vol%); #X obj 447 371 throw~ ch5; #X obj 515 371 throw~ ch6; #X obj 583 371 throw~ ch7; #X obj 651 371 throw~ ch8; #X msg 175 127 open 8ch_acheron_blasts.wav; #X msg 574 141 0; #X msg 667 238 \$1 3500; #X obj 667 259 unpack f f; #X obj 724 280 + 1025; #X text 719 238 fade time in ms; #X floatatom 724 301 5 0 0 0 - - -; #X obj 441 165 delay; #X obj 441 185 bng 15 250 50 0 empty empty empty 0 -6 0 10 -262144 -1 -1; #X obj 148 55 r acheron_play; #X obj 574 100 r acheron_fade; #X obj 148 76 t b a; #X obj 175 191 readsf~ 8 1e+12; #X obj 148 97 del 1500; #X connect 0 0 50 0; #X connect 1 0 50 0; #X connect 2 0 1 0; #X connect 3 0 8 0; #X connect 4 0 15 0; #X connect 5 0 16 0; #X connect 7 0 9 0; #X connect 10 0 12 0; #X connect 11 0 13 0; #X connect 14 0 39 0; #X connect 15 0 26 0; #X connect 16 0 26 0; #X connect 17 0 40 0; #X connect 18 0 11 1; #X connect 19 0 10 1; #X connect 20 0 7 1; #X connect 21 0 3 1; #X connect 22 0 34 0; #X connect 23 0 35 0; #X connect 24 0 36 0; #X connect 25 0 17 0; #X connect 26 0 27 0; #X connect 27 0 25 0; #X connect 28 0 37 0; #X connect 29 0 28 1; #X connect 30 0 24 1; #X connect 31 0 23 1; #X connect 32 0 22 1; #X connect 38 0 50 0; #X connect 39 0 26 0; #X connect 40 0 18 0; #X connect 40 0 19 0; #X connect 40 0 20 0; #X connect 40 0 21 0; #X connect 40 0 32 0; #X connect 40 0 31 0; #X connect 40 0 30 0; #X connect 40 0 29 0; #X connect 40 0 41 0; #X connect 41 1 42 0; #X connect 42 0 44 0; #X connect 44 0 45 1; #X connect 45 0 16 0; #X connect 45 0 46 0; #X connect 46 0 1 0; #X connect 47 0 49 0; #X connect 48 0 14 0; #X connect 48 0 45 0; #X connect 49 0 51 0; #X connect 49 1 38 0; #X connect 50 0 3 0; #X connect 50 1 7 0; #X connect 50 2 10 0; #X connect 50 3 11 0; #X connect 50 4 22 0; #X connect 50 5 23 0; #X connect 50 6 24 0; #X connect 50 7 28 0; #X connect 51 0 0 0; ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] how to avoid (most/many/some) readsf~ dropouts
Andy Farnell wrote: Greg, you may like to experiment with tweaking the disk caches. For linux see here: http://www.linuxdevcenter.com/pub/a/linux/2000/06/29/hdparm.html Thanks for the idea, Andy. It seems my stock Studio64 RT kernel is treating my SATA as a SCSI device -- but sdparm only coughs up errors. I'm gonna have to look deeper into this... G ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
