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/* (tabstops=8) ░░░░░░░░░░░░░░▒▒▒▒▒▒▒▒▒▒▒▒▒▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▒▒▒▒▒▒▒▒▒▒▒▒▒░░░░░░░░░░░░░░ ■ MOD Player Tutorial by FireLight ■ Copyright (c) Brett Paterson 1994-95 ■ ■ Last updated 16/6/95 ■ ░░░░░░░░░░░░░░▒▒▒▒▒▒▒▒▒▒▒▒▒▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▓▒▒▒▒▒▒▒▒▒▒▒▒▒░░░░░░░░░░░░░░ ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 0: ▓▒░ │ │ ░▒▓ Index ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ Section 1 : INTRODUCTION 1.1 Notes 1.2 Terminology 1.3 Contacting FireLight and feedback 1.4 Future versions Section 2 : THE LOADER 2.1 Notes 2.2 Verification 2.3 Load Module Name 2.4 Load Sample Information 2.5 Load Order Information 2.6 Load Pattern Data 2.6.1 Four bytes? 2.7 Load Sample Data 2.8 Phew :) Section 3 : PLAYING THE MOD 3.1 Ok Where Do I Start? 3.2 Setting The Timer's Speed 3.3 Player Logic 3.3.1 Orders/Patterns 3.4 Inside Update Row 3.5 Period Frequencies and Fine Tune 3.5.1 What do I do with this table? 3.5.2 Gravis UltraSound :) 3.6 Volume Section 4 : MISCELLANEOUS 4.1 Notes Without Instrument Numbers or Frequencies Section 5 : EFFECTS 5.1 Effect 0xy (Arpeggio) 5.2 Effect 1xy (Porta Up) 5.3 Effect 2xy (Porta Down) 5.4 Effect 3xy (Porta To Note) 5.5 Effect 4xy (Vibrato) 5.6 Effect 5xy (Porta + Vol Slide) 5.7 Effect 6xy (Vibrato + Vol Slide) 5.8 Effect 7xy (Tremolo) 5.9 Effect 8xy (Pan) 5.10 Effect 9xy (Sample Offset) 5.11 Effect Axy (Volume Slide) 5.12 Effect Bxy (Jump To Pattern) 5.13 Effect Cxy (Set Volume) 5.14 Effect Dxy (Pattern Break) 5.15 Effect Fxy (Set Speed) 5.16 Effect E0x (Set Filter) 5.17 Effect E1x (Fine Porta Up) 5.18 Effect E2x (Fine Porta Down) 5.19 Effect E3x (Glissando Control) 5.20 Effect E4x (Set Vibrato Waveform) 5.21 Effect E5x (Set Finetune) 5.22 Effect E6x (Pattern Loop) 5.23 Effect E7x (Set Tremolo WaveForm) 5.24 Effect E8x (Unused) 5.25 Effect E9x (Retrig Note) 5.26 Effect EAx (Fine Volume Slide Up) 5.27 Effect EBx (Fine Volume Slide Down) 5.28 Effect ECx (Cut Note) 5.29 Effect EDx (Delay Note) 5.30 Effect EEx (Pattern Delay) 5.31 Effect EFx (Invert Loop) Section 6 : APPENDIX - MOD FORMAT DOCUMENT ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 1: ▓▒░ │ │ ░▒▓ Introduction ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ ┌───────────────────┐ │ ░▒▓ 1.1 Notes ▓▒░ │ └───────────────────┘ New in this version: - Loader mistake fixed in section 2.6 -> - store SAMPLE_NUMBER as (byte1 AND 0F0h) + (byte2 SHR 4) should have been ^ - store SAMPLE_NUMBER as (byte0 AND 0F0h) + (byte2 SHR 4) ^ - Section 3.4, Inside Update row rewritten, the old one was weird and crap - Pattern break and pattern jump more accurately described - New section 2.6.1 - Four bytes? - Section 3.5.1 rewritten. Preamble: ========= I am covering the .MOD format here basically because it's not a very good idea to try and leap into a harder format like xm or s3m without prior knowledge. MOD still *IS* the most widely spread format so there's nothing wrong with coding a player for it. S3M is the next step up because it is basically just a wider .MOD with more octaves and a volume byte. (blah yeah I know there are 99 samples and more effects, that's just cosmetic though.) (ie s3m still use the same crap amiga frequencies as mod - for a PC format!). Assumptions: ============ Throughout the document, exaggerated length variable names are used, I don't actually use these sort of variable names but they help to make things clearer. eg "NUMBER_OF_PATTERNS". Variable names will be all stated in capitals. It is assumed you will have some sort of knowledge about - Sound Cards (and programming of sound cards, though I do include gus code in fmoda.asm) - Interrupt Handlers (I will cover this a bit though) Most of the time I present a type of pseudocode to try not to seem to biased towards a language, but some examples I have used straight C code only to demonstrate how I did it. C should be fairly intuitive to read so most people wont have that hard a time figuring it out. ┌─────────────────────────┐ │ ░▒▓ 1.2 Terminology ▓▒░ │ └─────────────────────────┘ TYPE LENGTH Bits RANGE BORLAND/TURBO C ──────────────────────────────────────────────────────────────────── byte 1 8 0-255 unsigned char word 2 16 0-65,535 unsigned int dword 4 32 0-4,294,967,295 unsigned long ──────────────────────────────────────────────────────────────────── Throughout this text I use the terms BYTE,WORD, and DWORD, to make the document more general to all languages. In C you can use typedefs to achieve the use of byte,word,dword terminology, and in pascal and asm the syntax is already suited to this anyway. ORDERS - orders are how the mod plays from 0 to length of song. PATTERNS - patterns are played in any ORDER, and are the physical information. TICK - I refer to a clock tick for the interrupt handler as a tick, some others use the term FRAME. I will be using the term tick throughout the whole document. ┌───────────────────────────────────────────────┐ │ ░▒▓ 1.3 Contacting FireLight and feedback ▓▒░ │ └───────────────────────────────────────────────┘ Contact is encouraged because I think I have left out some things and probably made some mistakes (not that I can see), and would like you to tell me about them. email : firelght@yoyo.cc.monash.edu.au post : Brett Paterson, 48/a Parr st, Leongatha, 3953, Victoria, Australia. phone : AU (056) 623795 IRC : FireLight on #coders, #trax or #aussies ┌─────────────────────────────┐ │ ░▒▓ 1.4 Future versions ▓▒░ │ └─────────────────────────────┘ Im really starting to get into this stuff, so here is what will appear in future versions of this document. o Mixing techniques - This is a very important section and I really want this to be included in here but need an experienced SB mod coder to write this section for me (anyone out there!!!) o How to handle multiple formats - talking about your internal format for handling multiple formats. I am currently updating fmod to support s3m and mtm and so info on these formats will be included. ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 2 : ▓▒░ │ │ ░▒▓ The Loader ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ ┌───────────────────┐ │ ░▒▓ 2.1 Notes ▓▒░ │ └───────────────────┘ Well first we've got to load the module in right? Following is a step by step way to code your loader, and storage issues will be discussed to help you along. I really don't feel like just writing another MOD format description, so you will find one in the appendix of section 6 written by lars hamre(?), the author of protracker. You WILL need to refer to the format document and this document side by side. The loader section of this document doesnt actually give a map of mod format and could be confusing, though it does go through it byte by byte. The following section has their subsections which are in boxes, and in each of these sections are 3 important subsections - EXPLANATION (describes what the section is on about, for understanding) - PSEUDOCODE (actually shows HOW to load the information) - STORAGE ISSUE (helps on how to store the information loaded) - SUGGESTION (a helpfull hint or suggestion to do after this step) I placed the pseudocode section before storage issues because I know you are probably going to be eager and want to jump into some code straight away. Storage issue follows just to be a guiding hand; not a 'must'. each pseudocode section follows on from the last. ┌──────────────────────────┐ │ ░▒▓ 2.2 Verification ▓▒░ │ └──────────────────────────┘ Explanation: ============ Before we attempt to load a mod, we should check that it is in fact a mod. Every mod has a unique signature, and in case of the .MOD format, this is in the form of a 4 letter string containing the letters "M.K.", or "8CHN" or a variety of other signatures for their mutated formats :) These describe the type of mod, and the identifier signature is stored at offset 1080 (438h) in the file, so should be checked first. PseudoCode: =========== - Seek to offset 1080 (438h) in the file - read in 4 bytes - compare them to "M.K." - if true we have a 4 channel mod - otherwise compare them to "6CHN" - if true we have a 6 channel mod - otherwise compare them to "8CHN" - if true we have an 8 channel mod - otherwise exit and display error message. There are also rare tunes that use **CH where ** = 10-32 channels Suggestion: =========== Use this point to store the number of channels in a variable of your choice (I just use a global variable called CHANNELS) ┌──────────────────────────────┐ │ ░▒▓ 2.3 Load Module Name ▓▒░ │ └──────────────────────────────┘ Explanation: ============ This is a trivial part of the loader and just holds the Title or name of the mod. It is the very first 20 bytes of the MOD. PsuedoCode: =========== - Seek back to position 0, the start of the file - read in 20 bytes, store as MODULE_NAME. Storage Issue: ============== The name of the module is a 20 byte string, padded by 0's. Here you can either store your module name as a global variable, in a character string, or do what I do and store all the general information about the mod in a structure like this struct MODHEADER { char NAME[20] ... other information (will get to this later) ... } MODHEAD OR just char NAME[20] It's a good idea to set up a structure like this for future use, there is a lot more infomration we will need to throw in here later, but of course you don't need a structure, you can keep it as a heap of loose variables :) And of course if you are not interested in displaying the name of the module you could just discard it. Suggestion: =========== Code a 1 line program to print out the name of your module to see if it's working properly. (exciting huh :) NOTE: The Module name is supposed to be padded by 0's, and terminated with a 0, but sometimes this is not the case. Sometimes a tracker will allow all 20 bytes to store characters, which means no NULL termintor byte. This causes functions like printf to give unpredictable output as it cannot find the NULL terminator. The way to fix this is just to use a loop and print out each character one at a time, or overwrite the 20th byte with a 0. ┌─────────────────────────────────────┐ │ ░▒▓ 2.4 Load Sample Information ▓▒░ │ └─────────────────────────────────────┘ Explanation: ============ Sample information is stored at the start of a MOD file, and contains all the relevant information for each of the 31 samples. This includes its name, length, loop points, finetune etc.. So from here we loop 31 times and read in a block of information about the sample according to the loop counter. PseudoCode: =========== - from this point, loop 31 times - for the sample # <loopcounter>.... - read in 22 bytes, store as SAMPLE_NAME - read in 2 bytes (word), store as SAMPLE_LENGTH * \ - read in 1 byte, store as FINE_TUNE @ /\ IMPORTANT: - read in 1 byte, store as VOLUME } see key - read in 2 bytes (word), store as LOOP_START * \/ below - read in 2 bytes (word), store as LOOP_LENGTH * / - end of loop KEY: * to get the real value in bytes, calculate it with (byte1*100h + byte2) * 2 @ for FINE_TUNE, if the value is > 7, subtract 16 from it to get the signed value (ie. 0-7 = 0-7, and 8-15 = -8 to -1) Storage Issue: ============== I think the best way to store information on the 31 instruments, is to store its information in a structure, then have an array of 31 of these intstrument structures. Like this : struct SAMPLE { char SAMPLE_NAME[22] word SAMPLE_LENGTH byte FINE_TUNE byte VOLUME word LOOP_START word LOOP_LENGTH (also some physical position information - see sample loading section. some possibilities are under GUS... dword GUS_OFFSET OR using main memory with sb say.. char *SAMP_BUFF (pointer to the actual physical data in memory) } now declare an array of 31 SAMPLEs. I do this in the general mod header structure which is explained fully in the next section. The other way which can be used is just to keep a heap of global arrays like this; char SAMPLE_NAME[31][22] word SAMPLE_LENGTH[31] byte FINE_TUNE[31] byte VOLUME[31] word LOOP_START[31] word LOOP_LENGTH[31] Suggestion: =========== Now code a little viewer once you have done this to make sure everything is stored properly. This is VERY a important step. Compare your output to the tracker it came from or a player that shows all sample information. ┌────────────────────────────────────┐ │ ░▒▓ 2.5 Load Order Information ▓▒░ │ └────────────────────────────────────┘ Explanation: ============ Ok now sample information is loaded, the next section of the module contains order information. Order information in a mod defines in what order patterns are going to be played. This means the composer could set orders 0 and 1 to pattern 0, for example, and the intent would be for pattern 0 to play twice. Its entry in the order table would look like this. ORDER : 0 1 2 3 4 5 6 7 8 9 PATTERN: 0 0 Note orders have to be from 0 to length of song, but patterns can be chopped and changed around in any order. The first byte from here will tell us the length of the song in -orders-, even though they are stored in 128 bytes of information. PsuedoCode: =========== - read a byte, store as SONG_LENGTH (this is the number of orders in a song) - read a byte, discard it (this is the UNUSED byte - used to be used in PT as the restart position, but not now since jump to pattern was introduced) Now we are at the orders table, this is 128 bytes long and contains the order of patterns that are to be played in the song. here we have to find out how many physical patterns there are in the module. How do we do this? Simple just check every order byte and the highest value found is stored as the number of patterns in the song. - set NUMBER_OF_PATTERNS to equal 0 - from this point, loop 128 times - read 1 byte, store it as ORDER <loopcounter> - if this value was bigger than NUMBER_OF_PATTERNS then set it to that value. - end of loop - read 4 bytes, discard them (we are at position 1080 again, this is M.K. etc!) Storage Issue: ============== One way is to go back to the other original MODhead structure, which contained general infomation about the mod. here is the entire structure. struct MODHEADER { char NAME[20] ; song name SAMPLE INST[31] ; instrument headers byte SONG_LENGTH ; song length byte NUMBER_OF_PATTERNS ; number of physical patterns byte ORDER[128] ; pattern playing orders } MODHEAD; or the second way would just to be store them all as global variables char NAME[20] ; song name byte SONG_LENGTH ; song length byte NUMBER_OF_PATTERNS ; number of physical patterns byte ORDER[128] ; pattern playing orders no array of samples here because if you saw the sample header loading section we just stored them all as their own arrays. Suggestion: =========== As always print out the 128 orders, and see if the pattern numbers displayed are correct. Now you should have a viewer that can just about display every bit of information about the module! OK that stuff was easy. Now it's time for something tougher.. the pattern data! ┌───────────────────────────────┐ │ ░▒▓ 2.6 Load Pattern Data ▓▒░ │ └───────────────────────────────┘ Explanation: ============ This is about the hardest part to code of the loader, and storage issues here are VERY important, so it will be discussed first. Im going to try and be as general as I can as I don't want to appear to be trying to steer you in any direction, but I will be specific enough to guide you. Storage Issues: =============== There are only a few ways to store pattern data, Ive spent some time pondering this issue. I found the only viable methods of storing pattern data are - 1- Linked List, using channels as nodes (dynamic but slow, well not THAT slow) 2- Fixed arrays (terribly memory wasting and messy) 3- Create and allocate a buffer the size we need to store all the patterns, and then use a roving pointer to access patterns later (sounds ok to me) Patterns really need to be stored DYNAMICALLY, or in other words only use as much memory as you need. Method 1 Was the method I used to begin with, for the sole reason that it is nicely dynamic and easy. It was quite ok to start on and was good enough for me (with GUS), but I scrapped that idea and went for the final method. Method 3 is much more general to all languages too. Method 2 Is out for this reason, it just isnt memory efficient enough. And also you cant subscript arrays in a normal high level language with indexes larger than 65536. (method 3 is an extension of this) Method 3. This method is quite easy and efficent to use and very dynamic, once you have worked out how to allocate and access huge pointers which can be up to 640kb big :) Players that seem to use this method are GUSPlay by Cascada, and ProTracker by Lars Hamre. FireMod 1.02 and higher, by myself uses this method. It works this way: - declare a pointer and allocate it the amount of memory calculated below; CHANNELS * 4 * 64 * (NUMBER_OF_PATTERNS+1) │ │ │ └─── (rows per channel) └─────── (bytes per note) Why add 1 to NUMBER_OF_PATTERNS? well because patterns start at 0, and finish at NUMBER_OF_PATTERNS, hence the aditional 1. If you didnt add 1 and there was only 1 pattern you would end up allocating 0 bytes for the pattern data :) This value is normally going to be a very big number, so a dword will be needed to store it. I initially had problems with data wrapping around at 64kb with my buffer using char far *, (say if it was 500kb large), but this was fixed by delcaring it with the huge keyword (look up online help to find out more) - eg : char huge *patbuff. So to find the physical pattern in your pattern buffer, calculate the offset with the formula (channels * 4 * 64) * pattno. Say we want to point to the start of pattern 4 in an 8CHN mod. (8 * 4 * 64) * 4 = 8192. So as you travel through this pattern just increment your pointer by 4 bytes at a time. A note is stored in the actuall file as 4 bytes, it is done in this fashion. The pseudocode below shows how to unravel this amigafied mess :) ┌─────────────────────────────────────┐ │ Byte 0 Byte 1 Byte 2 Byte 3 │ │─────────────────────────────────────│ │aaaaBBBB CCCCCCCCC DDDDeeee FFFFFFFFF│ └─────────────────────────────────────┘ aaaaDDDD = sample number BBBBCCCCCCCC = sample period value eeee = effect number FFFFFFFF = effect parameters PseudoCode: =========== - calculate amount of memory needed for NUMBER_OF_PATTERNS patterns like so: CHANNELS * 4 * 64 * (NUMBER_OF_PATTERNS+1) - create a base pointer and allocate the memory needed - From this point, loop for as many times as NUMBER_OF_PATTERNS - From this point, loop 64 * CHANNELS times (this equals 1 pattern) - read 4 bytes - store SAMPLE_NUMBER as (byte0 AND 0F0h) + (byte2 SHR 4) - store PERIOD_FREQUENCY as ((byte0 AND 0Fh) SHL 8) + byte1; - store EFFECT_NUMBER as byte2 AND 0Fh - store EFFECT_PARAMETER as byte 3 - increment pattern pointer by 4 bytes - end loop - end loop OK: === Alright so lets look at this again in simpler terms: - We have a big buffer that is meant to store all the pattern data - Then we start loading in the notes *4* bytes at a time, and unravel them into something meaningful as shown above. - store the new note variables one after the other, and it should fill the buffer to the exact size as was allocated in the beginning. Suggestion: =========== With EFFECT_PARAMTER, you might be tempted to store the 2 values stored in here as 2 seperate variables, eg. EFFECT_PARAMETER_X, and EFFECT_PARAMETER_Y. I used to store them this way but I assure you when you get into coding your effects this this method is quite inefficient, I saved memory and increased speed (but not noticably :) just by storing them in the 1 byte, and splitting them only in the few times that you do need it. (i.e, printing them out separately, or vibrato, or for finding out which E (extra) effect to use etc.) ┌───────────────────────────┐ │ ░▒▓ 2.6.1 Four bytes? ▓▒░ │ *IMPORTANT* └───────────────────────────┘ At this stage you're probably thinking.. how do I fit all this into only 4 bytes? For a start, DONT store the amiga periods as your note value. Convert each period to a note number. **See section 3.5.1** for more discussion on notes and frequencies. In summary you just scan through the amiga table until it matches the value you loaded in. Anyway even if you did store the amiga period value as your note (which you wont), then you can still fit it all into 4 bytes. The file did it so why cant you. I use bit allocation. This means I only use the bits I need in a byte, and not a whole byte. An example of this is the note volume is only capable of getting up to 64, so we only need 6 bits. The sample number goes up to 31. This only needs 5 bits. Follow here and see how things are allocated. This is similar to the way I do it in my player. In C you can allocate a variable and tell how many bits you want to use. In asm i'd say you would have to use a 4 bytes, and do the bit calculations yourself before you access them, which shouldnt be too hard. int note:11; // 0-?? = 11 bits = 0-2048 should be plenty for your needs. byte number:5; // 0-31 = 5 bits byte effect; // 0-15 = 4 bits, but use 8 to keep things even byte eparm; // 0-255 = 8 bits I actually use 3 bytes for my new player. I first convert finetunes to a middle C value in hz like s3m (see st3's tech.doc how this works), therefore I only need the amiga table for the actual notes and not the finetunes between. So I get something like note=7bits, number=5bits, effect=4bits, eparm=8bits, = 7+5+4+8 = 24 = 3bytes! ┌──────────────────────────────┐ │ ░▒▓ 2.7 Load Sample Data ▓▒░ │ └──────────────────────────────┘ PsuedoCode: =========== - From this point, loop 31 times - get the length of the sample # <loopcounter> (stored in your variable) At this point I use only GUS, and dump the sample to the GUS dram, but I could imagine if you were using Sound Blaster etc, you would just declare 31 pointers in memory and allocate them a SAMPLE_LENGTH sized buffer, then load the information into those buffers. When you need to play them you would mix the channels into a small buffer then DMA that buffer out to the sound card. - [SOUNDBLASTER] allocate a SAMPLE_LENGTH sized pointer to buffer in memory and load the sample into it - [DRAM-BASED-CARD (GUS)] poke/DMA bytes into DRAM, increment dword offset value GUS_OFFSET, and store that value next to the sample's information (along side length, volume, finetune, loop start etc) - check that your samples fit into (D)RAM, and exit with an error if they don't. - end loop ┌─────────────────────┐ │ ░▒▓ 2.8 Phew :) ▓▒░ │ └─────────────────────┘ Wasn't that bad was it? Now you have the FULL mod file stored away at your disposal, with samples ready to blast. Suggestions: ============ Now is a GOOD time to do some thorough testing. Do these things - Make sure your sample headers and information are stored correctly - Make sure your pattern data is stored perfectly.. it's quite important you know :) - Make sure your samples are stable in memory, and try to play them through your sound card.. you can have a few problems with misloaded samples I have found :) Also make sure the loop points are played correctly! - Make sure you deallocate your memory before quitting the program!! ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 3 : ▓▒░ │ │ ░▒▓ Playing the MOD ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ ┌─────────────────────────────────┐ │ ░▒▓ 3.1 OK where do I start ▓▒░ │ └─────────────────────────────────┘ I think the main thing you need to do now once you are satisfied your MOD is loaded properly, is to set up an interrupt function, and understand a bit about the way a MOD is played. Im going to use the system timer to hook onto here as an example, and if you want to use other interrupt servicers you can do that if you know how.. (ie GUS IRQ). You should know how to set up an interrupt handler yourself, but ill describe how to do it here with a bit of code to demonstrate. The system timer lies on INT 8 - Get the old handlers vector for int 8h, and store it away for later - Set your new handler function to the vector for int 8h ┌─────────────────────────────────────────────────────────────────────────┐ │ REMEMBER TO REHOOK YOUR OLD TIMER TO ITS ORIGNAL PLACE WHEN THE SONG IS │ │ FINISHED! │ └─────────────────────────────────────────────────────────────────────────┘ In C you would do that like this: oldhandler = _dos_getvect(8); setvect(8, handler); - where oldhandler has to have the prototype globally declared as void interrupt ( *oldhandler)(...); - for dummies the actuall handler function looks like this void interrupt modhandler(...) { // yes put 3 dots in here ... // do main loop here oldmodhandler(); // this is here to return int8 to what it // normally did. I'll crash without it. } In PASCAL it would look something like this GetIntVec($8, Addr(OldTimer)); SetIntVec($8, Addr(ModInterrupt)); - with the function looking something like (I have no idea if this is right as I don't do pascal) { $ F+,S-,W-} Procedure modhandler; Interrupt; Begin ... OldTimer; End; { $ F-,S+} If you're still not sure in C or pascal, check out the online manual on getvect/setvect etc.. ┌───────────────────────────────────────┐ │ ░▒▓ 3.2 Setting the timer's speed ▓▒░ │ └───────────────────────────────────────┘ Ok now your interrupt handler is already firing :) so one thing you must do is set it to the right speed, we don't want mods that play way to fast or slow, we want it at 125 BPM right now (or 50hz, or 50 ticks a second). How do you set the system timer's speed? if we want 50hz, we have to use a divisor to calculate the right rate like so. Speed = 1193180/50 <- 50 hz here, 1193180 is the divisor. mov dx, 0x43 mov al, 0x36 out dx, al mov dx, 0x40 mov ax, Speed <- here's the speed variable out dx, al shr ax, 8 out dx, al Now the interrupt function should be ticking away at 50 times a second. For other BPM's, which will be used because of the change tempo effect Fxy with values of 20h and up. If it is below 20h, then you change the SPEED and not the BPM. This is looked at later on. To convert BPM to HZ, you use : HZ = 2 * BPM / 5 (i.e 125bpm = 50hz) then SPEED = 1193180 / HZ for the set timer routine. Simple huh. You'll need this for effect Fxy, but don't worry about this until later. ┌──────────────────────────┐ │ ░▒▓ 3.3 Player Logic ▓▒░ │ └──────────────────────────┘ Now lets take a look at the interrupt function, this is where the playing is done. The SPEED of a song is the base on how your mod is played. Each row of a pattern is updated every SPEED number of clock ticks, so if a speed is 6, then you only update each row every 6 clock ticks. So on a speed like 3, the row is going to be updated every 3 ticks and will play twice as fast as speed 6. Inbetween you update certain tick sensitive effects, like portamentos, volume slides and vibrato. Diagramatically the playing of a mod looks like this. SPEED IS 6 tick# ┌─┘ 0: UPDATE ROW #0 <- update the 4,6 or 8 notes here in a mod's row. 1: --- \ 2: --- \ 3: --- >- certain effects are updated here 4: --- / 5: --- / 0: UPDATE ROW #1 1: --- 2: --- 3: --- 4: --- 5: --- 0: UPDATE ROW #2 etc.. Logically a very basic representation of playing a mod looks like this: ┌─────────────────────┐ │ STATIC TICK = SPEED │ - declaration, start it off at SPEED, not 0, as we │ │ want straight into the 'if tick >= speed condition' │ TICK = TICK + 1 │ - now increment the tick counter │ if TICK >= SPEED │ - if the tick # is bigger or equal than SPEED then │ update_row │ - update the CHANNEL number of notes for the new row │ tick =0 │ - reset tick to 0 │ ROW = ROW + 1 │ - incrememnt our row │ else update_effect │ - else we update the tick based effects. └─────────────────────┘ But you will have to take into account there are only 64 rows in a pattern, and if you hit 64 then jump to the next pattern and start at row 0 again. I say 64 because row 63's effects have to be played out before you jump to the next pattern. don't bother with update_effect for some time until you have got update_row going ok. ┌────────────────────────────────┐ │ ░▒▓ 3.3.1 Orders/Patterns ▓▒░ │ └────────────────────────────────┘ Just a short note on this. When you reach the end of the pattern or whatever, you need to go to the next order. Now say you had your order pattern numbers stored in an array as they should be, then it is simply a task of referencing that pattern number according to the index ORDER, and then repositioning your pattern pointer accordingly. ie. If your order list is something like this. ┌──────────────────────────┐ Order │ 0 1 2 3 4 5 6 7 8 9 .... │ │──────────────────────────│ Pattern │ 0 0 1 4 5 2 3 4 4 6 .... │ └──────────────────────────┘ and you have an array of patterns set up as ORDER_TABLE[128]. Selecting the appropriate pattern is as simple as finding ORDER_TABLE[ORDER]. To find the offset in your buffer you should know how to do by now by using some sort of formula like: offset = (CHANNELS * 4 * 64 * ORDER_TABLE[ORDER]) bytes, rows and to find the current row just add (CHANNELS * 4 * row). so the pattern+row formula ends up as : offset = (CHANNELS * 4 * 64 * ORDER_TABLE[ORDER]) + (CHANNELS * 4 * row). I calculate this figure before processing every row and set the pattern pointer, so that all I have to do is increment the row number or the order number and this formula will pick it up for me and set the pointer accordingly. ┌───────────────────────────────┐ │ ░▒▓ 3.4 Inside update row ▓▒░ │ └───────────────────────────────┘ Ok on every tick 0, we want to update CHANNELS number of channels PSEUDOCODE: ----------- - Point your note pointer to the correct offset in the pattern buffer, according to order and row Loop CHANNEL number of times { get NOTE from buffer get SAMPLE from buffer get EFFECT from buffer get EFFECT_PARAMETER from buffer if (SAMPLE > 0) then { LAST_INSTRUMENT[CHANNEL] = SAMPLE (we store this for later) volume[CHANNEL] = default volume of sample# SetVolume(volume[CHANNEL]) (actually do the hardware set here) } if (period >= 0) then { if (EFFECT does not = 3 and EFFECT does not = 5) then frequency[CHANNEL] = FREQ_TAB[NOTE + LAST_INSTRUMENT[CHANNEL]'s finetune] } (freq_tab[] should be your amiga frequency lookup table - see sec 3.5) (this line here is a bit of optimization for your player) if (effect# = 0 and parameter# = 0) then jump to SKIP_EFFECTS label ----- ----- PROCESS THE NON TICK BASED EFFECTS (see section 5 how to do this) ALSO GRAB PARAMETERS FOR TICK BASED EFFECTS (like porta, vibrato etc) ----- ----- label SKIP_EFFECTS: if (freqency[CHANNEL] > 0) then SetFrequency(frequency[CHANNEL]) if (period > 0 OR sample_offset > 00FFh) then { (Why 00FFh? because with sample offset anything below 1 * 100h is considered 0. See section 5.10 about this) if (vibratowavecontrol = retrig waveform) then { vibrato_position[CHANNEL] = 0 (see section 5.5 about this) vibrato_negative[CHANNEL] = 0 (see section 5.5 about this) } if (tremolowavecontrol = retrig waveform) then { tremolo_position[CHANNEL] = 0 (see section 5.8 about this) tremolo_negative[CHANNEL] = 0 (see section 5.8 about this) } PLAYVOICE * (here is gus biased, I guess for SB mixing you would mix in a section of the sample into a small buffer and dma it out here. You also have to take note if the sample is looping or not.. GUS does this for you of course ;) ) * (also remember to add the sample_offset value to the start of the sample begin address. If there was no sample offset then this value would be 0 and it would not affect the outcome.) } move pointer to next note in row (ie increment 4 bytes) } This is your main inner loop and the part that needs to be optimized. So make sure you can try and get it as fast as possible. *NOTE - setfrequency in this example is being passed amiga values, and should convert it to a relevant hardware value. ┌──────────────────────────────────────────────┐ │ ░▒▓ 3.5 Period Frequencies and Fine Tune ▓▒░ │ └──────────────────────────────────────────────┘ The formula for converting amiga period value to hz, is accomplished using ONE of the following formulas. Why there are 2 will be explained shortly. You are going to have to convert amiga frequencies to some sort of speed or frequency factor for YOUR sound card, so this part will show you how. PAL: 7093789.2 / (amigaval * 2) NSTC: 7159090.5 / (amigaval * 2) Say if we wanted to find the value in hz for middle note C-2. Looking up the amiga table we see the value for C-2 is 428 (see table below). therefore: PAL: 7093789.2 / (428 * 2) = 8287.14hz NSTC: 7159090.5 / (428 * 2) = 8363.42hz A quick explanation on PAL and NSTC. The amiga used to time its mods by sitting their interrupt handlers on the vertical retrace of the video screen so the period values they used in the tables are the amount of data to send to the amiga sound chip between interrupts, therefore changing the speed of data sent and the pitch of the note. Pretty stupid system huh. But I suppose back then they just wanted it to work and werent too worried about the future. Trackers like FastTracker 2 are taking a step in the right direction by using linear frequency tables.. ST3 took a step backwards by trying to base s3m on the mod format. This is MUSIC we are talking about not computer hardware. Which should I use? you are asking. Well I think the NSTC is the most widely accepted and used value, but it does not really matter. The only difference you might hear is a SLIGHT change in pitch, like about one fine tune out say. Inertia Play has a switch that lets you choose one or the other. Try flicking between the 2 while a song is playing to see what it is like. Here is a period table. This is straight out of protracker so it is bugfree, other tables you might see in like gusplay by cascada have bugs in it. Don't use it unless you can fix it. (ie the bug is about F-2 with finetune -3 or so.. FastTracker 1 has the bug try it out.) mt_PeriodTable ; Tuning 0, Normal dc.w 856,808,762,720,678,640,604,570,538,508,480,453 ; C-1 to B-1 dc.w 428,404,381,360,339,320,302,285,269,254,240,226 ; C-2 to B-2 dc.w 214,202,190,180,170,160,151,143,135,127,120,113 ; C-3 to B-3 ; Tuning 1 dc.w 850,802,757,715,674,637,601,567,535,505,477,450 ; same as above dc.w 425,401,379,357,337,318,300,284,268,253,239,225 ; but with dc.w 213,201,189,179,169,159,150,142,134,126,119,113 ; finetune +1 ; Tuning 2 dc.w 844,796,752,709,670,632,597,563,532,502,474,447 ; etc, dc.w 422,398,376,355,335,316,298,282,266,251,237,224 ; finetune +2 dc.w 211,199,188,177,167,158,149,141,133,125,118,112 ; Tuning 3 dc.w 838,791,746,704,665,628,592,559,528,498,470,444 dc.w 419,395,373,352,332,314,296,280,264,249,235,222 dc.w 209,198,187,176,166,157,148,140,132,125,118,111 ; Tuning 4 dc.w 832,785,741,699,660,623,588,555,524,495,467,441 dc.w 416,392,370,350,330,312,294,278,262,247,233,220 dc.w 208,196,185,175,165,156,147,139,131,124,117,110 ; Tuning 5 dc.w 826,779,736,694,655,619,584,551,520,491,463,437 dc.w 413,390,368,347,328,309,292,276,260,245,232,219 dc.w 206,195,184,174,164,155,146,138,130,123,116,109 ; Tuning 6 dc.w 820,774,730,689,651,614,580,547,516,487,460,434 dc.w 410,387,365,345,325,307,290,274,258,244,230,217 dc.w 205,193,183,172,163,154,145,137,129,122,115,109 ; Tuning 7 dc.w 814,768,725,684,646,610,575,543,513,484,457,431 dc.w 407,384,363,342,323,305,288,272,256,242,228,216 dc.w 204,192,181,171,161,152,144,136,128,121,114,108 ; Tuning -8 dc.w 907,856,808,762,720,678,640,604,570,538,508,480 dc.w 453,428,404,381,360,339,320,302,285,269,254,240 dc.w 226,214,202,190,180,170,160,151,143,135,127,120 ; Tuning -7 dc.w 900,850,802,757,715,675,636,601,567,535,505,477 dc.w 450,425,401,379,357,337,318,300,284,268,253,238 dc.w 225,212,200,189,179,169,159,150,142,134,126,119 ; Tuning -6 dc.w 894,844,796,752,709,670,632,597,563,532,502,474 dc.w 447,422,398,376,355,335,316,298,282,266,251,237 dc.w 223,211,199,188,177,167,158,149,141,133,125,118 ; Tuning -5 dc.w 887,838,791,746,704,665,628,592,559,528,498,470 dc.w 444,419,395,373,352,332,314,296,280,264,249,235 dc.w 222,209,198,187,176,166,157,148,140,132,125,118 ; Tuning -4 dc.w 881,832,785,741,699,660,623,588,555,524,494,467 dc.w 441,416,392,370,350,330,312,294,278,262,247,233 dc.w 220,208,196,185,175,165,156,147,139,131,123,117 ; Tuning -3 dc.w 875,826,779,736,694,655,619,584,551,520,491,463 dc.w 437,413,390,368,347,328,309,292,276,260,245,232 dc.w 219,206,195,184,174,164,155,146,138,130,123,116 ; Tuning -2 dc.w 868,820,774,730,689,651,614,580,547,516,487,460 dc.w 434,410,387,365,345,325,307,290,274,258,244,230 dc.w 217,205,193,183,172,163,154,145,137,129,122,115 ; Tuning -1 dc.w 862,814,768,725,684,646,610,575,543,513,484,457 dc.w 431,407,384,363,342,323,305,288,272,256,242,228 dc.w 216,203,192,181,171,161,152,144,136,128,121,114 * I personally used a sorted form of this table, that orders all the notes from C-1 with -8 finetune, then goes up through all the finetunes to B-3 with finetune +7. Makes things a lot easier I find. ┌─────────────────────────────────────────────┐ │ ░▒▓ 3.5.1 What do I do with this table? ▓▒░ │ └─────────────────────────────────────────────┘ I pondered this one myself for a bit when I first started. It would be nice if you could just store in the amiga values as your notes, then give them to your formula to use, and not even use a table to lookup amiga values. But there lies a problem. Namely finetune and arpeggio. If you have the amiga values stored as notes, then you will have no idea how much to fine tune according to the note you are on. If it was a linear table it would be fine (you would just say 'finetune = 2, so add 2 to the pitch'), but as it is actually a logarithmic table adding 2 on a C1 note gives a totally different tone to adding 2 on a C3 note. Forget storing the actual amiga periods as your notes, in your loader convert the periods to note numbers (see section 2.6.1), so you can use it to look up the period table later when the tune is playing. If you are still a bit confused this is how it is done. - Loading the pattern data, I looked up the amiga value loaded and gave it a number from 8 to 288. (36 notes, multiply it by 8 for finetunes between, remember each note is 8 finetunes apart, so it equals 288.) - start at 8 (C-1) because there are going to be 8 finetunes below C-1. - finish at 288 (B-3), and rememer there is going to be 7 finetunes above it. - You get this value by reading in the amiga value from the file, and scan through the period table (given above) until you find a match. (some trackers don't save the right numbers so I used a check if the number was between -2 to +2 from the actual value). Once you find the corresponding value, store the note as your (counter*8) where counter was the value you were incrementing as you went through the table. - Now the pattern data is loaded up with a nice linear set of notes. - when you actually play it just use your linear value as an index to look up the amiga table again to get the correct amiga period value. ok here's how I did it. ----------------------- period = ((byte0 & 0xF) << 8) + byte1; // read in the value from file current -> period = -1; // default value to -1, or 'nothing' for (count2=1;count2<37; count2++) { // start the search if (period > freqtab[count2*8]-2 && period < freqtab[count2*8]+2 ) current -> period = count2*8; // if found store the counter as } // the index for the note. If we went through the whole table and didnt find the value, then it is assumed there is no note, and it stays at -1. ┌────────────────────────────────────┐ │ ░▒▓ 3.5.2 Gravis UltraSound :) ▓▒░ │ └────────────────────────────────────┘ How to change to a GUS frequency??? Well you should find this sort of stuff yourself but because im gus biased ill talk a bit about it :) Assuming 44khz mixing rate: first : hz = 7159090.5 / ( amigaval * 2 ); next : gusfreq = ( hz / 44100) * 1024; simple huh.. the 44100 would change to whatever mixing rate you are using depending on the amount of voices. Ie say I use 20 voices so looking up this table... Frequency Active Voices 44100 14 or lower 41160 15 38587 16 36317 17 34300 18 32494 19 30870 20 29400 21 28063 22 26843 23 25725 24 24696 25 23746 26 22866 27 22050 28 21289 29 20580 30 19916 31 19293 32 ..My formula becomes gusfreq = ( hz / 30870 ) * 1024; BUT: with a bit of mathematical optimization I reduced this formula down to: hz = 7159090.50 / (freq * 2) gusfreq = ( hz / 30870 ) * 1024 = 3579545.25 / freq = hz / 30.14648438 now: gusfreq = ( 3579545.25 / freq ) / 30.14648438 = 118738.3894 / freq #define GUSfreq(x) 118738/x where x is the amiga value found in our period table.. saves a lot of calculation huh! (cuts 4 divs/muls down to 1 div) ok ok.. I know you are too lazy to work out 44.1khz, so it is 83117 / amigafreq. ┌────────────────────┐ │ ░▒▓ 3.6 Volume ▓▒░ │ └────────────────────┘ Handling volumes is one of the simplest parts of coding your player. It is just a matter of looking up a table or adjusting the percentage of the sample to be mixed into the final output. Remember there are actually 65 volume settings, just when you thought there were only 64 (040h) :). 0 is included which is absolutely no volume, and 64 is full volume. For gus users this is one of the best volume tables I have found anywhere. I have about 5 volume tables and this one is the one I use, it is quite loud but not so loud to cause bad clipping. Others I found are too soft. word GUSvol[] = { 0x1500, 0x9300,0xA900,0xB400,0xBC00,0xC180,0xC580,0xC980,0xCD80, 0xCF40,0xD240,0xD440,0xD640,0xD840,0xDA40,0xDC40,0xDE40, 0xDEF0,0xDFA0,0xE1A0,0xE2A0,0xE3A0,0xE4A0,0xE5A0,0xE6A0, 0xE7A0,0xE8A0,0xE9A0,0xEAA0,0xEBA0,0xECA0,0xEDA0,0xEEA0, 0xEEF0,0xEFE0,0xEF60,0xF1E0,0xF160,0xF1E0,0xF260,0xF2E0, 0xF360,0xF3E0,0xF460,0xF4E0,0xF560,0xF5E0,0xF660,0xF6E0, 0xF760,0xF7E0,0xF860,0xF8E0,0xF960,0xF9E0,0xFA60,0xFAF0, 0xFB70,0xFBF0,0xFC70,0xFCF0,0xFD70,0xFD90,0xFDB0,0xFDD0 }; ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 4 : ▓▒░ │ │ ░▒▓ Miscellaneous ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ This section describes some of the little things that should be taken note of when writing a mod player, but are VERY important. ┌─────────────────────────────────────────────────────────────┐ │ ░▒▓ 4.1 Notes Without Instrument Numbers or Frequencies ▓▒░ │ *IMPORTANT* └─────────────────────────────────────────────────────────────┘ This subsection is just about the most important of this whole section. Sometimes a composer will some seemingly strange methods to write a tune, i.e. leaving an instrument number off, or putting an instrument number but with no note! This part describes how to combat this. NO INSTRUMENT NUMBER: --------------------- C-2 01 C10 D-2 00 301 <- note no instrument number --- 00 300 You will notice, on the porta to note that the composer has left off the instrument number. Also notice that the previous note had the volume set to 10. Leaving off an instrument number causes the volume to stay as it is, and so the note slides, but still stays at volume 10. NO PERIOD VALUE OR NOTE: ------------------------ C-1 01 A07 --- 01 A07 <- no period value (note), but there are instrument numbers --- 01 A07 --- 01 A07 What this does is reset the volume on every note, and slides the volume down on every note too.. This gives a stuttering effect that is commonly used. It reinforces the last part (no instrument number), that if there is an instrument number, then the volume is reset to the sample's default volume. NOTE BUT NOTHING ELSE: ---------------------- C-1 01 000 D-1 00 000 E-1 00 000 This means the sample is reset to its starting position, on all 3 notes. CONCLUSION: ----------- - ONLY RESET VOLUME IF THERE IS AN INSTRUMENT NUMBER - ONLY RESET PITCH IF THERE IS A PERIOD VALUE/NOTE - ONLY RESET SAMPLE IF THERE IS A PERIOD VALUE/NOTE (and no effect 3xy, 5xy or EDx) ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 5 : ▓▒░ │ │ ░▒▓ Effects ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ This part of the document is one of the most sorely needed, it actually tells you HOW to code the effect, not just some vague reference on it and a basic explanation like I have seen in so many other docs. TERMINOLOGY: ============ Beside each effect, there are the 2 Y/N boxes.. these are; T0 : (TICK 0) This means the effect is updated or taken care of at the start of the row, or when the row is first encountered. INBETWEEN : This means the effect is updated on the other (speed-1) ticks that lie inbetween rows. When coding your player, go for effect Cxy first. It is the easiest and most substantial effect to enable. It will even make your tune resemble its normal self :). Then go for effect Fxy (set speed). ┌───────────────────────────────────┐ │ ░▒▓ 5.1 Effect 0xy (Arpeggio) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └───────────────────────────────────┘ This effect alternates the pitch rapidly to simulate a chord. It usually sounds very grating or harsh so it isnt used much except for chip tunes. EG: C-2 01 047 (I want to add to the pitch by 4 half notes then 7) Range: x = 1st semitone to add to note (0h-Fh) y = 2nd semitone to add to note (0h-Fh) so the effect 047 would generate a major, while effect 037 causes a minor. This is a tick based effect: Tick 0 Do nothing, Tick 1 you add the x arg, Tick 2 you add the y arg, Tick 3 you reset the frequency .... go back and do from tick 1 until we reach the next row You notice if SPEED is 1, then there will be no arpeggiation because there are no ticks inbetween. If SPEED is 2, then only the x arg is taken into account. Each note is 8 fine tunes apart, so use your finetune table to calculate the next row down if you like, or use a special arpeggio table to find the values to add. It is done something like this: - increment arpcounter by 1 - if arpcounter > 2 arpcounter = 0 - if arpcounter = 0 set the frequency to the normal value - if arpcounter = 1 set the frequency to the normal value + x # of finetunes - if arpcounter = 2 set the frequency to the normal value + y # of finetunes ┌───────────────────────────────────┐ │ ░▒▓ 5.2 Effect 1xy (Porta Up) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └───────────────────────────────────┘ This effect causes a pitch slide that goes up. EG: C-2 01 104 (I want to slide the frequency up 4 amiga values every tick) --- 00 104 (slide againt 4 values every tick) Range: xy = 00h-FFh You do this by resetting the frequency every tick, EXCEPT for the first one. The amount to slide by is the value given in EFFECT_PARAMETER You add the value to the AMIGA value of the frequency. Tick 0 Do nothing. Tick 1 add EFFECT_PARAMETER to the amiga frequency, and set it. Tick 2 add EFFECT_PARAMETER to the amiga frequency, and set it. Tick 3 add EFFECT_PARAMETER to the amiga frequency, and set it. .... keep going until end of note Remember B-3 is the highest note you can use, there is no law against sliding above it but it is not standard (some mods might be written thinking that the porta WILL stop at B-3, so be carefull). Personally I stop at 54, or approximately B-5. ┌─────────────────────────────────────┐ │ ░▒▓ 5.3 Effect 2xy (Porta Down) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └─────────────────────────────────────┘ This effect causes a pitch slide that goes down. EG: C-2 01 204 (I want to slide the frequency down 4 amiga values every tick) --- 00 204 (slide again 4 amiga values every tick) Range: xy = 00h-FFh You do this by resetting the frequency every tick, EXCEPT for the first one. The amount to slide by is the amound given in EFFECT_PARAMETER. You subtract the value from the AMIGA value of the frequency. Tick 0 Do nothing. Tick 1 subtract EFFECT_PARAMETER from the frequency, and set it. Tick 2 subtract EFFECT_PARAMETER from the frequency, and set it. Tick 3 subtract EFFECT_PARAMETER from the frequency, and set it. .... keep going until end of note Be careful you don't slide too low. Going below C-1 is non standard, and going below a frequency of 1 could cause horrible side effects :) ┌────────────────────────────────────────┐ │ ░▒▓ 5.4 Effect 3xy (Porta To Note) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [Y] └────────────────────────────────────────┘ This effect causes the pitch to slide towards the note specified. If there is no note specified it slides towards the last note specified in the Porta to Note effect. If no parameter use the last porta speed used for that channel. EG: C-2 01 000 D-2 01 301 (I want to set D-2 as the note to slide towards, and with a speed --- 00 300 of 1, then I just want to keep it sliding to D-2, and you already --- 00 300 know the speed so I wont bother telling you again) --- 00 300 Range: xy = 00h-FFh This effect can be buggy at first, but not too hard. on TICK 0: - If there is an argument given to the effect, then you must record that as PORTA_SPEED[channel]. (You need to remember all 4-8 channels worth of porta information - I have them as a global array) - If there is a note given, then you must store that as NOTE_TO_PORTA_TO[channel]. - But don't slide here, just like the other porta effects. - also, don't reset the note like you would normally if there was a frequency given (i.e. the D-2 in our example) On OTHER ticks: - Subtract or add PORTA_SPEED to the frequency (in AMIGA units), and set it. Subtract or add depending on if the current frequency is smaller or larger than NOTE_TO_PORTA_TO. ┌──────────────────────────────────┐ │ ░▒▓ 5.5 Effect 4xy (Vibrato) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └──────────────────────────────────┘ This effect causes the pitch to waver up and down around the base note. If no parameter use the last vibrato parameters used for that channel. EG: D-2 01 4A2 <- I want to vibrato the note D-2 with speed of A, and depth of 2 --- 00 400 <- Keep vibrating at A2 --- 00 4B3 <- now change to B3 --- 00 400 <- Continue vibrating at B3 Range: x = speed to vibrate at (0h-Fh) y = depth of vibrato (0h-Fh) This is simply a case of getting a sine table (the default wavecontrol - see section 5.20 for other vibrato wavecontrols), and following along it adjusting the frequency by adding or subtracting the value found according to the the position of the table, which is incremented by VIBRATO_SPEED. (ie you skip through the sine table VIBRATO_SPEED positions every tick) On TICK 0 the 2 vibrato values (position and neg flag) should be cleared to 0 if a new note is played, so we restart the waveform at the start again. Positioning vibrato pointer ---------------------------- There are 32 positions in the sine table. You want to ADD the values in the sinetable to the frequency, then once it gets to the end, you want to go back and SUBTRACT the same values from the frequency. This gives a nice wave. The reason we do this is because the sine table only contains half a wave (ie. a bump - see diagram). Running through it once then turning it upside down by negating it would produce a smooth running wave which oscillates up and down.. +1| **** /At this point we subtract from frequency | *** *** / Current 0 |**** ***|**** **** -> time | | *** *** -1| | **** 32 So once your VIBRATO_POS has gone past 32, then subtract 32 from it so it starts at a respectable place at the beginning again. THEN change the negation flag (ie flag: 0 for add values, 1 for subtract values). Sine Table ---------- This is the sine table used by Protracker. If a player calls itself fully protracker compatible, it really should be using this table. GUSPlay by Cascada uses a table that is slightly different, but I cant hear the difference :) 0, 24, 49, 74, 97,120,141,161, 180,197,212,224,235,244,250,253, 255,253,250,244,235,224,212,197, 180,161,141,120, 97, 74, 49, 24 Calculating depth ----------------- To calculate the amount or depth of the vibrato, you multiply the siner value by the effect parameter y, THEN you divide it by 128. Remember the divide by 128 (or shift right 7bits) must be implemented or you'll have some HUGE vibrato :) Setting the frequency. ---------------------- - Work out the size of the delta (delta means how much to add or subtract) - ie. delta = vibrato_depth[CHANNEL] * sine_table[vibrato_pos[CHANNEL] / 128 - if vibrato_negflag[CHANNEL] = 0, then SetFrequency(freq[CHANNEL]+delta) - else SetFrequency(freq[CHANNEL] - delta) Example code. ------------- For those interested this is how mine works, but I don't think it is 100% if (effect == 0x4 || effect == 0x6) { // work out the delta vib = vibdep[track]*sintab[vibpos[track]] >> 7; // >> 7 = div 128 // add the delta to the track's frequency if neg flag = 0 // subtract the delta to the track's frequency if neg flag = 1 if (vibneg[track] == 0) GUSSetFreq(track, GUSfreq(freq[track]+vib)); else GUSSetFreq(track, GUSfreq(freq[track]-vib)); vibpos[track]+=vibspe[track]; // increment vib position if (vibpos[track] > 31) { vibpos[track] -=32; // jump back to start if (vibneg[track]==0) vibneg[track] = 1; // change neg flag else vibneg[track]=0; } } ┌────────────────────────────────────────────┐ │ ░▒▓ 5.6 Effect 5xy (Porta + Vol Slide) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └────────────────────────────────────────────┘ This is a combination of Porta to Note (3xy), and volume slide (Axy). The parameter does not affect the porta, only the volume. If no parameter use the last porta to note parameter used for that channel. EG: C-1 01 000 D-1 01 301 <- start porta to note using speed of 3. --- 00 501 <- from here on keep doing porta, but slide volume down 1 as well. --- 00 501 --- 00 501 Range: x = amount to slide volume up by or (0h-Fh) y = amount to slide volume down by. (0h-Fh) This is exactly what it means, just do a 3xy first, then do a volume slide. The arguments only refer to the volume slide though and do not affect the porta. The porta is carried on from the last porta to note. So when you code your effect routine, it's like if (effect = 03h OR effect = 05h) DO_PORTA_TO_NOTE if (effect = 0Ah OR effect = 05h) DO_VOLUME_SLIDE kill 2 birds with 1 stone! ┌────────────────────────────────────────────┐ │ ░▒▓ 5.7 Effect 6xy (Vibrato+Vol Slide) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └────────────────────────────────────────────┘ This is a combination of Vibrato (4xy), and volume slide (Axy). The parameter does not affect the vibrato, only the volume. If no parameter use the vibrato parameters used for that channel. EG: C-1 01 4A2 <- start Vibrato with speed 0Ah, and depth 2. --- 00 601 <- from here on keep doing vibrato, but slide volume down 1 as --- 00 601 well. --- 00 601 Range: x = amount to slide volume up by or, (0h-Fh) y = amount to slide volume down by. (0h-Fh) This is exactly like effect 5xy, but just do a 4xy first, then do a volume slide. The arguments only refer to the volume slide though and do not affect the vibrato. The Vibrato is carried on from the Vibrato. So when you code your effect routine, it's like if (effect = 04h OR effect = 06h) DO_PORTA_TO_NOTE if (effect = 0Ah OR effect = 06h) DO_VOLUME_SLIDE kill 2 birds with 1 stone again! (hrmm thats 4 birds now :) ┌──────────────────────────────────┐ │ ░▒▓ 5.8 Effect 7xy (Tremolo) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └──────────────────────────────────┘ This effect causes the volume to oscillate up and down in a fluctuating style around the current volume, like vibrato but affecting volume not pitch. If no parameter use the last tremolo parameter used for that channel. EG: C-2 01 772 (I want to vibrate the volume up and down using speed 7 & depth 2) --- 00 700 (continue with the tremolo at 7,2) Range: x = speed to vibrate volume at (0h-Fh) y = depth of tremolo (0h-Fh) Seeing as this is a similar effect to vibrato, then we will use the same tables as it does. The only difference with tremolo is that you divide the delta (or deviation) by 64 and not 128. You also have to check for if the volume goes over or under 0 and 64. This means if the biggest value in the sine table 255 is divided by 64, then the biggest deviation with depth parameter of 1 would only be 4, on its peak. You're probably asking, what if the volume of the channel is 64? Well in this case you would only hear the negative side of the tremolo, when the volume dips down and then back to full. Same for the vice versa case if the volume is set to 0. On TICK 0 the 2 tremolo values (position and neg flag) should be cleared to 0 if a new note is played, so we restart the waveform at the start again. This is how it works. - Work out the size of the delta (delta means how much to add or subtract) - ie. delta = tremolo_depth[CHANNEL] * sine_table[tremolo_pos[CHANNEL] / 64 if tremolo_negflag[CHANNEL] = 0, then { check if volume[CHANNEL] + delta > 64 and clip delta accordingly SetVolume(volume[CHANNEL]+delta) } else { check if volume[CHANNEL] - delta < 0 and clip delta accordingly SetVolume(volume[CHANNEL] - delta) } - increase tremolo_position pointer and set neg flag accordingly (For any more information check vibrato because they really are the same. It is explained in more detail, and the sine table mentioned is stored in there also.) ┌──────────────────────────────┐ │ ░▒▓ 5.9 Effect 8xy (Pan) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └──────────────────────────────┘ This effect is non-Protracker, but is worth mentioning. It was introduced by Otto Chrons in DMP (dual mod player), and causes the left/right position of the current channel to be set to the position specified. Hence Panning. EG: --- 00 800 (Set the position of the channel to the far left) 00 = far left 40 = middle 80 = far right A4 = surround * (* Surround is usually achieved by having 2 copies of the sample, 1 inverted, and you play them at -exactly- the same time, with one of the pair panned fully left, and the other (the inverted one say) panned fully right. This will give a surround effect. If you play both the samples in the same pan position they will cancel each other out. Experiment with this in a tracker. Using GoldWave(tm) you can invert a sample. As efffect 8xy is a channel command, you will have to in effect have 2 channels (voices) ready for this channel, and make sure you set one voice to the full left, and the other inverted, and to the full left. You CAN have surround sound on a GUS.) ┌─────────────────────────────────────────┐ │ ░▒▓ 5.10 Effect 9xy (Sample Offset) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └─────────────────────────────────────────┘ This effect causes the note to start playing at an offset into the sample, instead of just from the start. It is used so that the beginning of a sample is not played, but skipped. EG: C-2 01 942 (I want to start the note playing at 4200h bytes into the sample) Range: xy = 00h-FFh As seen in the example, the argument is the first 2 digits of a 4 digit number (in hex) that the offset should take place from. so SAMPLE_OFFSET = EFFECT_PARAMETER * 0100h What you do to enable this effect is when you tell your soundcard or mixing buffer the start of the sample, also add to it the value SAMPLE_OFFSET and then play it. Quite simple really. Remember to check if the user set an offset that is larger than the sample! ┌────────────────────────────────────────┐ │ ░▒▓ 5.11 Effect Axy (Volume Slide) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └────────────────────────────────────────┘ This effect causes the volume of the track to slide up or down. EG: A-2 01 A01 <- slide the volume down 1 * (speed-1) units --- 00 A01 <- slide the volume down 1 * (speed-1) units --- 00 A01 <- slide the volume down 1 * (speed-1) units --- 00 A20 <- now slide the volume up 2 * (speed-1) units --- 00 A20 <- slide the volume up 2 * (speed-1) units Range: x = amount to slide volume up by or, (0h-Fh) y = amount to slide volume down by. (0h-Fh) On this affect you either slide the volume up x, or down y, but not both. This is a tick based effect so should be processed once a tick but not tick 0. if x > 0 then slide volume up x if y > 0 then slide volume down y if x > 0 and y > 0 then do nothing. On tick 0: Take note of the volume slide, but do nothing On other ticks: if x > 0 then add x to volume[CHANNEL] and set the volume if y > 0 then subtract y to volume[CHANNEL] and set the volume * before setting the volume, make sure you havent slid past 0 or 64. ┌───────────────────────────────────────────┐ │ ░▒▓ 5.12 Effect Bxy (Jump To Pattern) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └───────────────────────────────────────────┘ This effect jumps to a specified channel (in hex) EG: --- 00 B10 (I want to jump to order 10h, or 16) Range: xy = 00h-FFh This effect is fairly simple, after the ticks for the note are finished, then reset the position of the order, starting at row 0 again. Make sure you don't jump over the end of the song length, and if you do then set it to the last order. * if you increment your row after your PlayNote() function, then row should be set to -1, so it is 1 less than 0, then as the tick handler adds 1 to the row it is 0 again, and nothing is wrong. ┌──────────────────────────────────────┐ │ ░▒▓ 5.13 Effect Cxy (Set Volume) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └──────────────────────────────────────┘ This effect sets the volume of a channel. EG: C-2 01 C20 (I want to set the volume of the channel to 20h) Range: xy = 00h-40h This is about the easiest and first effect you should code. It is just a simple case of setting the tracks volume to the argument specified (in hex) The volume cannot be set past 40h, and if it is then set it to 40h. Only process this effect on tick 0, and likewise only set the volume on tick 0. ┌─────────────────────────────────────────┐ │ ░▒▓ 5.14 Effect Dxy (Pattern Break) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └─────────────────────────────────────────┘ This effect breaks to the next pattern starting at the specified row. EG: --- 00 D32 (I want to break from this pattern and start at row 32h on the next pattern) Range: xy = 00h-3Fh (0-63 decimal) This effect is similair to effect Bxy or pattern jump. You only jump to the next pattern though, and you start tracking again at the specified row. The row should not be bigger than 63, and if it is take it as 0. It works something like this: - increment order (only once, some mods have more than 1 pbreak on a row which could cause an increment order twice or more!) - set row to be x*10 + y. (we have to get the decimal value not the hex) * if you increment your row after your PlayNote() function, then row should be set to (x*10+y -1), so it is 1 less, then as the tick handler adds 1 to to the row again, nothing is wrong. ┌─────────────────────────────────────┐ │ ░▒▓ 5.15 Effect Fxy (Set Speed) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └─────────────────────────────────────┘ This effect sets the speed of the song or the BPM. EG: --- 00 F07 (I want to set the speed of the song to 7 ticks a row) --- 00 F7D (I want to set the bpm of the song to 125 or 7Dh) Range: xy = 00h-1Fh for speed xy = 20h-FFh for BPM This has 2 parts to it. If the user specifies a parameter from 0 - 1Fh, then it is just simply a case of setting your speed variable, otherwise you need to set your bpm variable and reset the timer speed. This is demonstrated in section 3.2 on how to change the speed of the system timer according to beats per minute. ┌──────────────────────────────────────┐ │ ░▒▓ 5.16 Effect E0x (Set Filter) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └──────────────────────────────────────┘ This effect turns on or off the hardware filter (not applicable to most pc sound cards) EG: --- 00 E01 (I want to turn the filter on) --- 00 E00 (I want to turn the filter off) Range: x = 0 to turn hardware filter off, 1 to turn it on (0-1) There isnt much to say about this effect, except for that it is a hardware function which was designed to turn on the amiga's filter. If you wanted to you could try implementing this effect in the SBPro's h/w filter. ┌─────────────────────────────────────────┐ │ ░▒▓ 5.17 Effect E1x (Fine Porta Up) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └─────────────────────────────────────────┘ This effect slides the pitch up by x amiga value's per row. EG: C-2 01 E11 (I want to start at note C-2, and move pitch up one amiga value) --- 00 E11 (keep sliding up...) --- 00 E11 Range: x= amount to slide up by. (0h-Fh) This effect is only processed once per row, on tick 0, and it is as simple as just subtracting x from the current channel's frequency. (remember you subtract to raise the pitch.) You don't subtract any finetunes or anything, just do a straight subtraction of x from the amigaval. ┌───────────────────────────────────────────┐ │ ░▒▓ 5.18 Effect E2x (Fine Porta Down) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └───────────────────────────────────────────┘ This effect slides the pitch down by x amiga value's per row. EG: C-2 01 E21 (I want to start at note C-2, and move pitch down one amiga value) --- 00 E21 (keep sliding down...) --- 00 E21 Range: x = amount to slide pitch down by. (0h-Fh) This is identical to effect E2x, except but you add to the amigaval of the channel by x, and don't subtract. ┌────────────────────────────────────────────┐ │ ░▒▓ 5.19 Effect E3x (Glissando Contrl) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └────────────────────────────────────────────┘ This effect causes a change in the effect 3xy (porta to note). It toggles whether to do a smooth slide or whether to slide in jumps of semitones. EG: --- 00 E31 (I want to turn on Glissando and have portas slide in semitones) --- 00 E30 (I want to turn off Glissando and have portas slide smoothly) Range: x = 0 to turn off glissando, 1 to turn it on (0-1) By default this value should be set as 0, or doing a smooth slide. It is achieved by adding or subtracting the desired porta value too or from the amiga value in effect 3xy, but you already knew that :). With glissando turned on it is a different story. It is just simply a case of setting the frequency to the next highest semitone (or 8 finetune values) if you are sliding the pitch up, and vice versa for going down. To implement this just keep a gliss flag and check it while doing your porta effect in your UpdateEffect function. ┌────────────────────────────────────────────┐ │ ░▒▓ 5.20 Effect E4x (Vibrato Waveform) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └────────────────────────────────────────────┘ This effect set the waveform for the vibrato command to follow. EG: --- 00 E42 (I want to select the squarewave function for the vibrato command) --- 00 E40 (I want to select the default sinewave for the vibrato command) Range: x = vibrato function to select (0-7) The following values of x select its corresponding vibrato function x=0 : Set sine wave (default) x=1 : Set Ramp Down |\|\|\ _ _ x=2 : Set Squarewave |_| |_| |_ x=3 : Set Random (anywhere) x=4 : don't retrig Sine waveform x=5 : don't retrig RampDown waveform x=6 : don't retrig Squarewave waveform x=7 : don't retrig random waveform - Sine wave is covered in the vibrato section (5.5), just apply a sine wave to the frequency. - Square wave is simply subtracting and adding the VIB_DEPTH*256 (then divided by 128) to the current frequency, alternating the add/subtract every VIB_SPEED number of ticks. - retrig waveform means that you start the vibrato waveform from position 0 everytime a new note is played. If you have set the wave control flag to 4 or more, then the waveform is not restarted, and just continues from the previous position in the vibrato waveform. ┌────────────────────────────────────────┐ │ ░▒▓ 5.21 Effect E5x (Set Finetune) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └────────────────────────────────────────┘ This effect sets the finetune on a selected instrument. EG: --- 01 E5F (I want to set the finetune of instrument 1 to -1) Range: x = value of finetune to set (0h-0Fh) if the value is > 7, just subtract 16 from it to get the signed value. (ie. 0-7 = 0-7, and 8-15 = -8 to -1) This effect is really easy, and I don't know why more players support it, apart from it being a useless effect :). To implement it, just - check the instrument number - get the finetune value in the effect - set the finetune for that instrument. ┌────────────────────────────────────────┐ │ ░▒▓ 5.22 Effect E6x (Pattern Loop) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └────────────────────────────────────────┘ This effect allows the user to loop a part of a pattern x number of times. EG: C-2 01 E60 (I want to set the loop start at this point) --- 00 000 --- 00 E64 (I want to loop back to the starting point 4 times) Range: x=marks loop starting point, or sets the number of times to loop to the starting point (0h-0Fh) This effect is done in the following fashion. - If parameter x = 0, note down the row number - if parameter x > 0, then - if PATTERN_LOOP = 0, then set PATTERN_LOOP = x else PATTERN_LOOP = PATTERN_LOOP -1 - if PATTERN_LOOP > 0 row = stored row number. (if we are still looping then jump back) Remember when declaring the PATTERN_LOOP variable to initialize it as 0. Jumping back should just be a matter of setting your row number to the stored pattern loop number, and once the row is finished it should start playing at the specified position again. This is how my function works, in the UPDATE_NOTE function, or handler for tick 0. case 0x6 : if (eparmy == 0) patlooprow = row; // store position of param=0 else { if (patloopno == 0) patloopno=eparmy; // set times if 0 else patloopno--; // else subtract 1 if (patloopno > 0) row = patlooprow-1; // if looping do jump } ┌────────────────────────────────────────────┐ │ ░▒▓ 5.23 Effect E7x (Tremolo WaveForm) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └────────────────────────────────────────────┘ This effect set the waveform for the tremolo command to follow, just like vibrato. EG: --- 00 E42 (I want to select the squarewave function for the tremolo command) --- 00 E40 (I want to select the default sinewave for the tremolo command) Range: x = tremolo function to select (0-7) The following values of x select its corresponding tremolo function x=0 : Set sine wave (default) x=1 : Set Ramp Down |\|\|\ _ _ x=2 : Set Squarewave |_| |_| |_ x=3 : Set Random (anywhere) x=4 : don't retrig Sine waveform x=5 : don't retrig RampDown waveform x=6 : don't retrig Squarewave waveform x=7 : don't retrig random waveform see section 5.20 for information. ┌──────────────────────────────────────────┐ │ ░▒▓ 5.24 Effect E8x (16 pos panning) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └──────────────────────────────────────────┘ This effect lets you do 16 position panning EG: --- 00 E80 (I want to set the channel's pan value to the far left) --- 00 E8F (I want to set the channel's pan value to the far right) Range: x=position to pan too (0h-0Fh) On tick 0, just read in the parameter and set the relative panning value for the channel. ┌───────────────────────────────────────┐ │ ░▒▓ 5.25 Effect E9x (Retrig Note) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └───────────────────────────────────────┘ This effect retiggers the current note every x ticks. EG: C-2 01 E93 (I want to retrig the note every 3 ticks - at speed 6 this would --- 00 000 retrig it only once) C-2 01 E91 (I want to retrig the note every tick - at speed 6 this would retrig the note 5 times) Range: x=ticks between retriggers (0h-0Fh) On this effect you need to use the modulus operator to check when the retrig should happen. If x is 1 say, then it should retrig the note SPEED number of times in one note. ie. tick MOD 1 = 0 always, so you would be retrigging every note. tick MOD 2 = 0 on even numbers, 1 on odd numbers, so you would be retrigging every other note. etc. When it does happen just play out the note as you would normally. The note is played on tick 0 as it would normally be. ┌────────────────────────────────────────────┐ │ ░▒▓ 5.26 Effect EAx (Fine VolSlide Up) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └────────────────────────────────────────────┘ This effect slides the volume up x values per row. EG: C-2 01 C00 (I want to start at note at volume 0) --- 00 EA1 (Now I want to slide the volume up for the channel by 1 unit) --- 00 EA1 (keep sliding up by 1 unit...) Range: x= amount to slide up by. (0h-Fh) This effect is only processed once per row, on tick 0, and it is as simple as just adding x to the current channel's volume. It is only processed on tick 0, and is not touched at all in the other ticks. The only checking to be done is for volumes larger than 64. hint: for all these volume commands, only do the checking for bounds once, just before you actually set the volume. ┌──────────────────────────────────────────────┐ │ ░▒▓ 5.27 Effect EBx (Fine VolSlide Down) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └──────────────────────────────────────────────┘ This effect slides the volume up x values per row. EG: C-2 01 EB1 (I want to slide the volume down for the channel by 1 unit) --- 00 EB1 (keep sliding down by 1 unit...) --- 00 EB1 (keep sliding down by 1 unit...) Range: x= amount to slide up by. (0h-Fh) This effect is only processed once per row, on tick 0, and it is as simple as just subtracting x from the current channel's volume. It is only processed on tick 0, and is not touched at all in the other ticks. The only checking to be done is for volumes smaller than 0. hint: for all these volume commands, only do the checking for bounds once, just before you actually set the volume. ┌────────────────────────────────────┐ │ ░▒▓ 5.28 Effect ECx (Cut Note) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └────────────────────────────────────┘ This effect cuts the volume of the note to 0 after x amount of ticks. EG: (at speed 6 say) C-2 01 EC3 (I want to stop the note at tick 3, or half way between 2 notes) Range: x= number of ticks to wait before zeroing samples volume. (0h-Fh) This effect is ignored on tick 0, but on tick x when you are updating tick based effects, then just set the volume of the channel to 0. Of course if the user specified x as a number more than the speed of the song, then it would be ok because it would never get to tick x, and the effect is ignored. ┌──────────────────────────────────────┐ │ ░▒▓ 5.29 Effect EDx (Delay Note) ▓▒░ │ UPDATED: T0 [N] : INBETWEEN [Y] └──────────────────────────────────────┘ This effect waits for x amount of ticks before it actually plays the sample. EG: (at speed 6 say) C-2 01 ED4 (I want to delay playing this note for another 4 ticks) Range: x= number of ticks to wait before playing sample. (0h-Fh) This effect is ignored on tick 0, AND you must make sure you don't play the sample on tick 0. When you arrive at tick x then just play the sample as you would normally. Again if the user specified x as a number more than the speed of the song, then it would be ok because it would never get to tick x, and the effect is ignored. ┌─────────────────────────────────────────┐ │ ░▒▓ 5.30 Effect EEx (Pattern Delay) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └─────────────────────────────────────────┘ This effect delays the pattern for the time it would take to play x number of notes. EG: C-2 01 EE8 (I want to play the c-2 note then wait for 8 notes before.. C-2 01 000 ... playing the next note) Range: x= number of notes to delay pattern for. (0h-Fh) To implement this effect you are going to have to modify your main interrupt handler (see section 3.3): You are going to have to keep a counter that is subtracted every time your SPEED number of ticks is up, but don't play the note. You must still keep playing the effects though. It would look something like this. if (tick >= speed) { ... blah blah blah etc... if (patdelay == 0) { increment row. playnote. } else patdelay --; } else doeffects This just boils down to not playing the note or incrementing the row for x number of notes, until the pattern delay counter is 0. When it is 0 the mod should keep playing as if nothing had happened. ┌───────────────────────────────────────┐ │ ░▒▓ 5.31 Effect EFx (Invert Loop) ▓▒░ │ UPDATED: T0 [Y] : INBETWEEN [N] └───────────────────────────────────────┘ This effect inverts a sample loop or plays it backwards. EG: C-2 01 EF4 (I want to play the loop in this sample backwards at speed 4) Range: x = speed to set invert loop at (0h-0Fh) This effect is not supported in any player or tracker. Don't bother with it. ┌────────────────────────────────────────────────────────────────────────────┐ │ ░▒▓ : SECTION 6 : ▓▒░ │ │ ░▒▓ Protracker 1.1B Format Document ▓▒░ │ └────────────────────────────────────────────────────────────────────────────┘ Offset Bytes Description 0 20 Songname. Remember to put trailing null bytes at the end... Information for sample 1-31: Offset Bytes Description 20 22 Samplename for sample 1. Pad with null bytes. 42 2 Samplelength for sample 1. Stored as number of words. Multiply by two to get real sample length in bytes. 44 1 Lower four bits are the finetune value, stored as a signed four bit number. The upper four bits are not used, and should be set to zero. Value: Finetune: 0 0 1 +1 2 +2 3 +3 4 +4 5 +5 6 +6 7 +7 8 -8 9 -7 A -6 B -5 C -4 D -3 E -2 F -1 45 1 Volume for sample 1. Range is $00-$40, or 0-64 decimal. 46 2 Repeat point for sample 1. Stored as number of words offset from start of sample. Multiply by two to get offset in bytes. 48 2 Repeat Length for sample 1. Stored as number of words in loop. Multiply by two to get replen in bytes. Information for the next 30 samples starts here. It's just like the info for sample 1. Offset Bytes Description 50 30 Sample 2... 80 30 Sample 3... . . . 890 30 Sample 30... 920 30 Sample 31... Offset Bytes Description 950 1 Songlength. Range is 1-128. 951 1 Well... this little byte here is set to 127, so that old trackers will search through all patterns when loading. Noisetracker uses this byte for restart, but we don't. 952 128 Song positions 0-127. Each hold a number from 0-63 that tells the tracker what pattern to play at that position. 1080 4 The four letters "M.K." - This is something Mahoney & Kaktus inserted when they increased the number of samples from 15 to 31. If it's not there, the module/song uses 15 samples or the text has been removed to make the module harder to rip. Startrekker puts "FLT4" or "FLT8" there instead. Offset Bytes Description 1084 1024 Data for pattern 00. . . . xxxx Number of patterns stored is equal to the highest patternnumber in the song position table (at offset 952-1079). Each note is stored as 4 bytes, and all four notes at each position in the pattern are stored after each other. 00 - chan1 chan2 chan3 chan4 01 - chan1 chan2 chan3 chan4 02 - chan1 chan2 chan3 chan4 etc. Info for each note: _____byte 1_____ byte2_ _____byte 3_____ byte4_ / / / / 0000 0000-00000000 0000 0000-00000000 Upper four 12 bits for Lower four Effect command. bits of sam- note period. bits of sam- ple number. ple number. Periodtable for Tuning 0, Normal C-1 to B-1 : 856,808,762,720,678,640,604,570,538,508,480,453 C-2 to B-2 : 428,404,381,360,339,320,302,285,269,254,240,226 C-3 to B-3 : 214,202,190,180,170,160,151,143,135,127,120,113 To determine what note to show, scan through the table until you find the same period as the one stored in byte 1-2. Use the index to look up in a notenames table.