bosone

i received the reply from alesis: **** I apologize for the delay in responding. Prices are as follows: n.1 - AL3201B $5.75/ea n.1 - AL3101 $3.95/ea (AL3102 ) $3.80/ea n.1 - AL1101 $1.75/ea n.1 - AL1201 $1.95/ea n.1 - AL1402 $4.28/ea n.1 - AL1401A $4.00/ea The AL3201B is back ordered and I will not have available stock unitl late May. The AL1201 is currently out of stock but I am expecting a delivery by the 15th of April. ***

i didn't understood well what was the matter!!! i'm impatient to know something more! ;D

i found this on harmony central review. i think it could be interesting! :) ***** This is an interview with SID chip Designer. It shows what capabilities the C=64 HAS. Did Commodore ever plan to build an improved successor to the SID ? I don't know. After I left I don't think there was anyone there who knew enough about music synthesis to do much more than improve the yield of the SID chip. I would have liked to have improved the SID chip before we had to release to production, but I doubt it would have made any difference to the success of the Commodore 64. Can you give us a short overview of the SIDs internal architecture ? It's pretty brute-force, I didn't have time to be elegant. Each "voice" consisted of an Oscillator, a Waveform Generator, a Waveform Selector, a Waveform D/A converter, a Multiplying D/A converter for amplitude control and an Envelope Generator for modulation. The analog output of each voice could be sent through a Multimode Analog Filter or bypass the filter and a final Multiplying D/A converter provided overall manual volume control. As I recall, the Oscillator is a 24-bit phase-accumulating design of which thelower 16-bits are programmable for pitch control. The output of the accumulator goes directly to a D/A converter through a waveform selector. Normally, the output of a phase-accumulating oscillator would be used as an address into memory which contained a wavetable, but SID had to be entirely self-contained and there was no room at all for a wavetable on the chip. The Sawtooth waveform was created by sending the upper 12-bits of the accumulator to the 12-bit Waveform D/A. The Triangle waveform was created by using the MSB of the accumulator to invert the remaining upper 11 accumulator bits using EXOR gates. These 11 bits were then left-shifted (throwing away the MSB) and sent to the Waveform D/A (so the resolution of the triangle waveform was half that of the sawtooth, but the amplitude and frequency were the same). The Pulse waveform was created by sending the upper 12-bits of the accumulator to a 12-bit digital comparator. The output of the comparator was either a one or a zero. This single output was then sent to all 12 bits of the Waveform D/A. The Noise waveform was created using a 23-bit pseudo-random sequence generator (i.e., a shift register with specific outputs fed back to the input through combinatorial logic). The shift register was clocked by one of the intermediate bits of the accumulator to keep the frequency content of the noise waveform relatively the same as the pitched waveforms. The upper 12-bits of the shift register were sent to the Waveform D/A. Since all of the waveforms were just digital bits, the Waveform Selector consisted of multiplexers that selected which waveform bits would be sent to the Waveform D/A. The multiplexers were single transistors and did not provide a "lock-out", allowing combinations of the waveforms to be selected. The combination was actually a logical ANDing of the bits of each waveform, which produced unpredictable results, so I didn't encourage this, especially since it could lock up the pseudo-random sequence generator by filling it with zeroes. The output of the Waveform D/A (which was an analog voltage at this point) was fed into the reference input of an 8-bit multiplying D/A, creating a DCA (digitally-controlled-amplifier). The digital control word which modulated the amplitude of the waveform came from the Envelope Generator. The Envelope Generator was simply an 8-bit up/down counter which, when triggered by the Gate bit, counted from 0 to 255 at the Attack rate, from 255 down to the programmed Sustain value at the Decay rate, remained at the Sustain value until the Gate bit was cleared then counted down from the Sustain value to 0 at the Release rate. A programmable frequency divider was used to set the various rates (unfortunately I don't remember how many bits the divider was, either 12 or 16 bits). A small look-up table translated the 16 register-programmable values to the appropriate number to load into the frequency divider. Depending on what state the Envelope Generator was in (i.e. ADS or R), the appropriate register would be selected and that number would be translated and loaded into the divider. Obviously it would have been better to have individual bit control of the divider which would have provided great resolution for each rate, however I did not have enough silicon area for a lot of register bits. Using this approach, I was able to cram a wide range of rates into 4 bits, allowing the ADSR to be defined in two bytes instead of eight. The actual numbers in the look-up table were arrived at subjectively by setting up typical patches on a Sequential Circuits Pro-1 and measuring the envelope times by ear (which is why the available rates seem strange)!

i found this on harmony central review. i think it could be interesting! :) ***** This is an interview with SID chip Designer. It shows what capabilities the C=64 HAS. Did Commodore ever plan to build an improved successor to the SID ? I don't know. After I left I don't think there was anyone there who knew enough about music synthesis to do much more than improve the yield of the SID chip. I would have liked to have improved the SID chip before we had to release to production, but I doubt it would have made any difference to the success of the Commodore 64. Can you give us a short overview of the SIDs internal architecture ? It's pretty brute-force, I didn't have time to be elegant. Each "voice" consisted of an Oscillator, a Waveform Generator, a Waveform Selector, a Waveform D/A converter, a Multiplying D/A converter for amplitude control and an Envelope Generator for modulation. The analog output of each voice could be sent through a Multimode Analog Filter or bypass the filter and a final Multiplying D/A converter provided overall manual volume control. As I recall, the Oscillator is a 24-bit phase-accumulating design of which thelower 16-bits are programmable for pitch control. The output of the accumulator goes directly to a D/A converter through a waveform selector. Normally, the output of a phase-accumulating oscillator would be used as an address into memory which contained a wavetable, but SID had to be entirely self-contained and there was no room at all for a wavetable on the chip. The Sawtooth waveform was created by sending the upper 12-bits of the accumulator to the 12-bit Waveform D/A. The Triangle waveform was created by using the MSB of the accumulator to invert the remaining upper 11 accumulator bits using EXOR gates. These 11 bits were then left-shifted (throwing away the MSB) and sent to the Waveform D/A (so the resolution of the triangle waveform was half that of the sawtooth, but the amplitude and frequency were the same). The Pulse waveform was created by sending the upper 12-bits of the accumulator to a 12-bit digital comparator. The output of the comparator was either a one or a zero. This single output was then sent to all 12 bits of the Waveform D/A. The Noise waveform was created using a 23-bit pseudo-random sequence generator (i.e., a shift register with specific outputs fed back to the input through combinatorial logic). The shift register was clocked by one of the intermediate bits of the accumulator to keep the frequency content of the noise waveform relatively the same as the pitched waveforms. The upper 12-bits of the shift register were sent to the Waveform D/A. Since all of the waveforms were just digital bits, the Waveform Selector consisted of multiplexers that selected which waveform bits would be sent to the Waveform D/A. The multiplexers were single transistors and did not provide a "lock-out", allowing combinations of the waveforms to be selected. The combination was actually a logical ANDing of the bits of each waveform, which produced unpredictable results, so I didn't encourage this, especially since it could lock up the pseudo-random sequence generator by filling it with zeroes. The output of the Waveform D/A (which was an analog voltage at this point) was fed into the reference input of an 8-bit multiplying D/A, creating a DCA (digitally-controlled-amplifier). The digital control word which modulated the amplitude of the waveform came from the Envelope Generator. The Envelope Generator was simply an 8-bit up/down counter which, when triggered by the Gate bit, counted from 0 to 255 at the Attack rate, from 255 down to the programmed Sustain value at the Decay rate, remained at the Sustain value until the Gate bit was cleared then counted down from the Sustain value to 0 at the Release rate. A programmable frequency divider was used to set the various rates (unfortunately I don't remember how many bits the divider was, either 12 or 16 bits). A small look-up table translated the 16 register-programmable values to the appropriate number to load into the frequency divider. Depending on what state the Envelope Generator was in (i.e. ADS or R), the appropriate register would be selected and that number would be translated and loaded into the divider. Obviously it would have been better to have individual bit control of the divider which would have provided great resolution for each rate, however I did not have enough silicon area for a lot of register bits. Using this approach, I was able to cram a wide range of rates into 4 bits, allowing the ADSR to be defined in two bytes instead of eight. The actual numbers in the look-up table were arrived at subjectively by setting up typical patches on a Sequential Circuits Pro-1 and measuring the envelope times by ear (which is why the available rates seem strange)!

ehi... what do you mean!?!?!? do you have in mind an analogue/digital multiFX unit!??!?!?!?!?!?!?

is there someone who has the old midibox16 with the port of firmaware for midibox64 for using the sequencer?? i tried, but i have soim rpoblem in running the sequencer... maybe i have to read better the tutorial, but for now, when the seq "starts" i hear no notes at all!!

is there someone who has the old midibox16 with the port of firmaware for midibox64 for using the sequencer?? i tried, but i have soim rpoblem in running the sequencer... maybe i have to read better the tutorial, but for now, when the seq "starts" i hear no notes at all!!

i tried to burn the 1.909 version downloaded some time ago, but the behaviour remains the same... the queation is: when i change the layer in my midibox16 in snap mode, should appear the arrows in the display, showing the pot positions?? this do not happen i tried both with merger on (and masterkeyb in midibox in) and merger off (and midibox in connected to my soundcard midi ouput, with soundcard input connected to soundcard output)... i have an idea about bypassing this problem with a short modification of the code, taking a "snapshot" of the setup just before bankswitch and restoring it just after.... maybe i can learn PIC programming!

it seems that you can request free samples from cirrus logic for the chips you used (CD8405A instead of 02). so, can you please post me your schematic for you converter?? i'd like to try it!! you can write pvt at m.bosi(at)inwind.it thank you very much!

look at www.ebay.de for LCD display!! i buyed one for 5 euro+shipping (i'm in italy, shipping costed 2 euro!) unbelievable!

i have the old midibox 16 (v1.908 i think), and just today i mounted the LCD screen. everything works perfect, but when i set the "snap" or "relative" or "parallax" mde nothing happens!!! it seems that the pot behaviour is always set to "normal". the behaviour is as following: i switch to "snap" mode i set layer 1 and all pots are at 0 i set layer 2 and all pots are at 0 now i turn back to layer 1 and move pot 1 to 64. when i set layer 2, the pot 1 is at 64!! (even in the display!!!) i don't see the "arrows" that are supposed to indicate pot position before the "snap"... the only moment in which i see the arrows in the display is aftr startup... what can be? ??? i have another question: if i burn the latsest midibox HP firmware in my PIC even with the old midibox16 hardware and PCB will it work? thanks!!

after hearing my sid at work i wonder: how do you deal witha "stereo sid"? ok, lets suppose i build another core+sid module, and then? i think the only thing to do is connecting both unit to the same midi in and then "play them together" with the same patch. should the two SID produce slightly different sounds for a "stereo" effetc? other ideas?

YEAH! today arrived the free samples from microchip and my SID unit returned back to life!!! now i only miss the case, then i'll be able to groove!

ehi, MrMusicman can you send us the schematics and the price of your unit?

how many patches can handle the SID without the bankstick??? for my lates try i could not achieve program change in SID (even after having transferred all the default bank with the java librarian). is it correct?? will the program changes work once i will get the bankstick??

this look very interesting! just for the sake, i wrote a mail to alesis semiconductor to see if it is possible to buy small quantity of those chips... let's see! it could be interesting for me building a small sfpid/analog converter!!! :-)

now i'm *quite* happy because i have seen on FedEx site that on monday 24 they should arrive the PIC samples from microchip... i cannot wait to get'em... and i hope that SID will still work!!

you are very polite, but i'm waiting for the free microchip samples.... otherwise, i don't think the 4MHz versions will work, since i have a 20MHz oscillator... thank you!

i'm SOOOOOOOOOOOO sad...... :( this afternoon i finished my SID. i was very happy! everything worked at first try and i succeded in listening to the old c64 sounds! cute! the PCB were not in case, i was just testing. i even added a led to see when the unit is turned on. after the tests, i left all on my workbech, and unplugged the power supply. then, after some hours i decided to make some order. i take the core module and ZACK! a spark! the led lighted for a while and then nothing........ i reinsterted the plug, but the led did not lit, and there was no signs of life. the unit made NO sounds!!!!!!!! i unplugged the PIC, tunred on the power and made the voltage test: evrerything was correct, and the led was lit... i tried to insert the PIC into the JDM programmer... nothing!!! there were NO data on it and it was not possible to burnb the firmware... my PIC is DEAD!!!!!!!!!!!! R.i.P. :-((( now i have to wait for the microchip free samples, and i suppose they will arrive on late april... the BIG question is now: is the SID dead, also??? how can i control if it is sane??????? SID and CORE module are connected to the same power supply and were linked altogether when happened the spark...

yeah! i've just ordered the free pic from microchip! unbelievable! ;D

you are absolutely right.. sorry... :P however, i don't think i will never take myself to upgrade my midbox 16 to a midibox 64 with faders and rotary econders and so on.. now i have just found a LCD on german ebay (i hope it will work when i will mount it) and all i want is to have access to the "snap mode" of my pots. however, i think i have understood that for SID it should be much better... damn that from distrelec (where i ordered all the things for SID and CORE module) they had not the new generation PIC!! now i have to find it in another way!

??? ??? OK, everyone is speaking about the MIOS and the new PIC series... but what are they, precisely??? in another topic replyed me that using the new PIC and mios with my new-build SID module i can add pots to control the synth in realtime with "hardware" knobs. well... but everything i have to is change my brand new PIC 16F877-20 with the new PIC series??? no changes to the PCB and to the components??? i'm sorry if is a very dummy and stupiud question!! :-[

se qualcuno ha bisogno di PCB per i progetti di thorsten, posso mettervi in contatto con chi ha fatto le mie!

i'm in trouble with two SID module cap. on the schematic, between pin 1-2 and 3-4 of the sid there are two caps of 470pF. in the other drawing (the one with the components mounted on the PCB) those caps are signed as 1nF... :o is it an error?? ??? for now, i mounted 2x 470nF...

i wonder if it is possible to connect three knobs to the basic SID design (core+sid module) in order to control the volume (mosto important), filter cutoff and resonance...