details:

'pure data' means pure data vanilla or pure data extended. it excludes the use of any other sound processing application to make or sequence sounds, and excludes the use of VST, ladspa, audio unit etc plugins within pure data itself. triggering and controlling sounds with external hardware such as midi knobs and faders, arduino, etc, is totally fine. using an audio editor to record and do simple edits (crop, normalize, etc) of pd material is also fine, as long as the sound is not resequenced and no effects are added.

basically the sound needs to start and end in pd. this doesn't mean it all has to happen realtime. i have been having lots of fun lately with running sampled pd sounds through various sampling devices and effects)

my goal at the moment is to make some 'sample packs' of pure pd sounds which i will then release here and in other places under the creative commons license, so any submissions would need to be covered by an equally free agreement.

after that, what i do with those samples, and what anyone else does with those samples is up to us, and would be covered by creative commons. my final goal is to make an album worth of music and sounds using only pure data,

i can be contacted by email at hard.off@gmail.com

So I am new to PD and am taking a class. I currently have an assignment due in a few days and I am trying to finish it. I have to design a patch with multiple subpatches that will do to things.

Layer 2 On this layer the sound will play at a normal speed/pitch and it will play every 10 seconds. Gradually, the periodicity of the repetition will start increasing and it will take 2 minutes to end up playing every 5 seconds. After that minute the processes will start again, over and over. In other words, there will be an accelerando in the repetition of the unmodified sound. 2 points Layer 3 On this layer the sound will play over and over but the speed/pitch will change randomly so sometimes the sound will play faster (up to a tenth of a second) and sometimes the sound will play slower (up to 10 seconds). When the file plays faster it will naturally play at a higher pitch, and when it plays slower it will play at a lower pitch. On each repetition the sound should play sometimes forward and sometimes backwards.

Any suggestions on how to do this would be I would like to recieve because I am really confused

HI,
i'm working on an installation based on this apllication made in java

i communique with pd via OSC
for each collision pd receive a bang with two parameters
height tube
position tube

i'm looking for synthesis metal bars sounds to transform this "thing" into a musical instrument
there is samples here
http://obiwannabe.co.uk/html/sound-design/sound-design-audio.html
http://obiwannabe.co.uk/sounds/effect-clonk-002-bar.mp3
http://obiwannabe.co.uk/sounds/effect-clonk-002-bar.mp3
http://obiwannabe.co.uk/sounds/effect-clonk-004-iron.mp3
http://obiwannabe.co.uk/sounds/effect-clonk-006-bar.mp3

What kind of simple patch should i have to make for this goal?

au revoir
Denis

Hey guys,

Wikipedia these , ad netsearch - I had for my memoire a few years ago...

Walt Disney- Fantasia_ they toured with live mixers in surround for that film. They used the pan pots developed by Blumlein.

(Brit)Blumlein was the guy who said mono can imitate spac eby being panned between two distribution sources (speakers).

There was Fletcher (Am)- Mr WALL OF SOUND -
each source, a speaker... Too expensive - and impossible for mass distrib. Blum wins...

Nowadays, with PD...

Surround
dac 1 2 3 4 5 6 7 8 then link with adat 9 10 11 12 etc....

(Soundcards - Protools digi01, motu 828 now sell for under 300 used!! - Motut drivers are current, but ot for linux. RME is linux/win/mac ready)

(Sound on Sound has great articles on the following!!!)

if you follow dolby...
123 1 and 3 ambient front, 3 center voice
4 5 - left right behind, mix of front with 5-20 ms delay, depending on image, with high and low cuts, so it imitates how we hear sound when it is behind us.
6 Bass - dumped to woofer below 110-80 hz.

Haven't read up on DTS (check sound on sound)

all the numbers are different depending on your setup of your program of choice, and sound card..But the distro is l/r front, l/r rear with delay ad filter cut. Center front, and bass.

This creates a uniform distro method for cinema - good to know if you have mixed a film stereo, and do not want the dolby effect in place!!!
or, when you get a dolby mixer to work on your film, she or he is preparig your film for this in the cinema...

test it:
just do a surround mix, saving your work. play it back on multitrack through a dolby amp from a surround dvd with coaxial...

then connect to the amp from a dvd reader with each track connected by a coaxial cable...

then connect a multitrack sound card directly to your amp, with each track linked to an out.....

you should hear some differences...

But the man at IEM, Mr PD and GEM, has been working on their surround for years - and has distributed the patch recently. Plus ambison is around for max...and I thought, there was a PD port....

But it begs the question of mass distro versus unique design and experiences of sound....

Hello,

I previously wrote a topic as sort to understand how to detect motion coming from a webcam and play in consequence a soundfile. I wanted to play that sound and not to break its reading until its finished.

I got an good answer here :

http://puredata.hurleur.com/sujet-1486-webcam-detection-play-sound-entirely

Now, I have a more complicated issue.
Imagine we have four people coming in a room. Each time somebody is passing the door, then a sound is red. If these four individuals entered the room too fast, then the soundfile would be re-played several times in a short time and we couldn't hear it entirely. That's why I'll use a spigot connected to a delay box (answer given above).

However, imagine these four people come in the room close together. They'd hear a single sound until it ends. After, there would be nothing unless that a new person would come into the room or quite these people get out the room.

To be more simple, the idea is that everyone has a sound given. If 10 individuals are coming, then it'll have 10 sounds played following each.

Please, would you have any advice to give me as sort to stock, keep numbers of entries in the room and then when a sound is finished run the next one ?
Many thanks !

Something to share that combines a few different models in a linked way.

Start with a wind model based on turbulence, objects in the path vary their signals according to wind speed and their size and texture.

http://www.obiwannabe.co.uk/sounds/effect-wind3.mp3

And a rain model with carefully distributed droplets that make little clicks according to a range of textures they hit...

http://www.obiwannabe.co.uk/sounds/effect-plainrain.mp3

Next is a window pane built around a square lamina with glass-like character Here's a few knocks on the virtual window with a virtual stick.

http://www.obiwannabe.co.uk/sounds/effect-knockonwindow.mp3

and finally I combine them all in the same auditory scene with causal linkage, so the rain lashes against the window...

http://www.obiwannabe.co.uk/sounds/effect-rainywindow.mp3

(Total object count 80 operators)
Andy

below you'll find my lsmod info. echomixer, the alsa-toolkit utility for echo audio products did work after doing [ # alsaconf ] however, I tried to test my config simply by doing this;

# aplay -vv *
ALSA lib confmisc.c:670:(snd_func_card_driver) cannot find card '0'
ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device
ALSA lib confmisc.c:391:(snd_func_concat) error evaluating strings
ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device
ALSA lib confmisc.c:1070:(snd_func_refer) error evaluating name
ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device
ALSA lib conf.c:3968:(snd_config_expand) Evaluate error: No such device
ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default
aplay: main:550: audio open error: No such device

So therer is still a missing piece.

Module Size Used by
snd_layla24 36356 0
snd_seq_oss 40084 0
snd_seq_midi 9792 0
snd_seq_midi_event 8160 2 snd_seq_oss,snd_seq_midi
snd_seq 60456 5 snd_seq_oss,snd_seq_midi,snd_seq_midi_event
snd_rawmidi 28992 2 snd_layla24,snd_seq_midi
snd_seq_device 9708 4 snd_seq_oss,snd_seq_midi,snd_seq,snd_rawmidi
firmware_class 11744 1 snd_layla24
snd_pcm_oss 52032 0
snd_mixer_oss 20704 1 snd_pcm_oss
snd_pcm 91396 2 snd_layla24,snd_pcm_oss
snd_timer 26500 2 snd_seq,snd_pcm
snd 65908 9 snd_layla24,snd_seq_oss,snd_seq,snd_rawmidi,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
soundcore 11204 1 snd
snd_page_alloc 11304 2 snd_layla24,snd_pcm

Dunno what snd_pcm is returning there, but you should see a separate driver for the Layla24
like this

$ lsmod

snd-seq-midi 5152 0 (unused)
snd-virmidi 2080 0
snd-seq-virmidi 5128 0 [snd-virmidi]
snd-seq-midi-event 6240 0 [snd-seq-midi snd-seq-virmidi]
snd-seq 48784 0 [snd-seq-midi snd-seq-virmidi snd-seq-midi-event]
snd-layla24 149732 3 <--------*here*
snd-pcm 85860 2 [snd-layla24] <---and pcm is using it

You don't have to recompile kernel or anything, find a driver and use insmod

Apparently there's a utils package at Alsa Project website for the Echo Layla24 that sets up everything. Have you tried that one?

also, there's an ALSA Wiki up now that may help you

Well for now I'm concerned about getting an authentic sound without worrying too much about emulating the specific operation of the hardware. I also want to add a few extras that the original didn't have, like vibrato, sweeping of the triangle channel, and maybe some other small odds and ends.

The pulse channel sound is simple to emulate, especially if you aren't concerned about timing their length & envelope data against other components, like the frame counter or interrupt lines. The triangle is a bit trickier to get authentic.

The noise channel is particularly difficult to emulate, at least for the inexperienced like me. The NES noise sound in itself is easy to reproduce as a sample using 4bit level quantized noise. The 2A03 actually uses a long shift register and a XOR gate to generate a new pseudo-random bitstream for noise samples. Rather than use my very own enveloping like I did for pulse and triangle channels, I will have to reproduce the native specs of the counters/timers and decay envelope modes, especially to get the looped-decay noise channel mode to sound authentic.

So I guess I will be using some of the same dataflow and control logic that the hardware uses, but I want to cut as many corners as I can right now, especially where I can easily provide userdata through the GUI instead of poking 6502 assembly. Then I can use my own, simpler methods for programmable manipulation of all of the inputs, but ideally get the same-sounding output as I would programming the actual hardware.

Right now I'm going through this document to try and get a full picture of the hardware:
http://nesdev.parodius.com/NESSOUND.txt

I believe that has everything needed to directly emulate the channels, I just gotta keep studying the hell out of it until I can determine all of the specifics on timing, mode switching, sample sizes and such.

The zenpho patch looks similar to what I want to use eventually for making real music, I'll probably refer to that a few times. I see he uses a completely diferent PWM routine than I do. Once I get the NES channels sounding properly, I plan to keep adding voices from other old sound chips I enjoy along with more extras and use it as my main synth.

Thanks alot for that headlessbarbie link. Really amazing stuff. I've had thoughts about later on trying to emulate the 2A03 hardware directly, so that I possibly could put pd on a board with a fast CPU (maybe a SuperH) that would be small enough to fit in a NES cart. Then I could use pd as just an interpreter between the user and the real live sound hardware.

@daisy said:

I have read some where that "if a voice is at same pitch and same loudness and still if one recognize that two voices are different , it is becuase of TIMBRE (tone quality)". (I agree there are other features as well who need to consider).

Timbre is another word for spectrum. The spectrum of a sound is the combination of basic sine waves that are mixed together to make it. Every sound (except a sine wave) is a mixture of sine waves. You can make any sound by adding the right sine waves together. This is called synthesis.

@daisy said:

First Question:
So how we can calculate the TIMBRE of voice? as fiddle~ object is used to determine the pitch of voice? what object is used for TIMBRE calculation?.

[fft~] object splits up the spectrum of a sound. Think of it like a prism acting on a ray of light. Sound which is a mixture of sines, like white light, goes in. A rainbow of different colours comes out. Now you can see how much red, blue, yellow or green light was in the input. That's called analysis.

So the calculation that gives the spectrum doesn't return a single number. Timbre is a vector, or list of numbers which give the frequencies and amplitudes of the sine waves in the mixture. We sometimes call these "partials".

If you use sine wave oscillators to make a bunch of new sine waves and add them together according to this recipe you get the original sound back! That's called resynthesis.

@daisy said:

Second Question:
And how one can change TIMBRE? as pitch shifting technique is used for pitch? what about timbre change?

Thanks.

Many things change timbre. The simplest is a filter. A high pass filter removes all the low bits of the spectrum, a bandpass only lets through some of the sine waves in the middle, and so on...

Another way to change timbre is to do analysis with [fft~] and then shift some of the partials or remove some, and then resynthesise the sound.

@daisy said:

I have a kind of general idea (vcoder). but how to implement it? and how to change formant?.

A vocoder is a bank of filters and an analysis unit. Each partial that appears in the analysis affects the amplitude of a filter. The filter itself operates on another sound (often in real time). We can take the timbre of one sound by analysing it and get it to shape another sound that is fed through the filters. The second sound takes on some of the character of the first sound. This is called cross-synthesis.

/doc/4.fft.examples/05.sheepgoat.pd
Help -> 7.Stuff -> Sound file tools -> 6.Vocoder