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Digitized Voice Programmer's Toolkit for the PC ----------------------------------------------- Version 2.0 Copyright (c) 1988,1989,1990,1991 Farpoint Software Toolkit Instructions ------------------------------------------------------------------------------ *************************************************************************** * * * As this documentation file is being written, it has been almost two * * and a half years since Version 1 was released, and three and a half * * years since DIGITS.ARC and HIDI.ARC. I have listened to a fair * * amount of user feedback by this time, and feel that I know a little * * more about what people want in a product like this one. This release * * is an attempt to respond to this feedback. Here are some improvements * * you will find from the last version: * * * * (1) BETTER SOUND QUALITY. Everyone wants better sound. No surprise * * here. This toolkit provides a dramatic improvement in output * * quality over version 1. This goal took priority over everything * * else. If you have version 1, try an A/B comparison. * * * * (2) DIGITIZER HARDWARE IS AVAILABLE. Many users preferred not to * * build the digitizer themselves; unfortunately, Farpoint Software * * never manufactured digitizers for commercial sale, making do-it- * * yourself the only option. I'm sure a number of potential users * * were turned away by this. We have now fixed this problem, at a * * significant financial risk. (So buy one!) * * * * (3) SIMPLER PROGRAMMING INTERFACE. The complexity of using the core * * assembly language routines in high-level programs has been * * reduced as much as possible. We have also included a true device * * driver which allows any program written in any language to output * * voice in a manner identical to a simple file write operation. * * * * And, for balance, here are the things that went away since version 1: * * * * (1) Background operation and file read-on-the-fly. This was * * necessary in order to achieve the desired sound quality * * improvement. You can't allow interrupts when sub-microsecond * * timing is required. * * * * (2) Small voice data files. They got bigger. Eight times as big when * * uncompressed, four times when using Mu-Law compression. * * * * * * Many thanks to all the honest people who registered for version 1. * * Everyone who registered for version 1 is now automatically registered * * for version 2. * * * * - Alan D. Jones * * * *************************************************************************** Introduction ------------ This toolkit is a combination of software and hardware designed for the purpose of mechanizing and simplifying the process by which programmers may create digitized voice recordings, store them on disk, edit the voice data files, and incorporate digitized voice playback into their own high-level language programs. The recording of digitized voice requires an external hardware device. This archive includes the schematic of the device in several files, each appropriate for one popular printer type. The device is also available directly from Farpoint Software. Note that it is not the same as the digitizer used with DVPT version 1. This new digitizer had to be a little more complex to accomplish its task. Playback of the digitized voice, however, requires NO SPECIAL HARDWARE. The sound is produced with the built-in speaker provided in nearly all PC's and PC-compatible machines. This means that programs may be written for general distribution which will play voice messages on the user's machine as it exists. Here is a list of the major features of the current software package: (1) Operates under the DOS environment. (2) Provides a full set of voice record/playback control routines which are directly callable from many high-level languages including C and Pascal. They are also, of course, callable from assembly language. (3) There are no length limitations on either the size of the memory buffers or the size of the voice data files on disk other than the physical limits of the machine itself. 64k is not a special number. (4) A sophisticated voice data file editor is provided. It has moved away somewhat from the "tape recorder" model, and now provides a file indexing feature that can be used to directly create "sound dictionary" files of a format that is relatively easy for the programmer to manipulate. (5) Several short example programs are included, written in C, which demonstrate the use of the calls to the voice modules, both as linked object modules as a device driver. Shareware Notice ---------------- The Digitized Voice Programmer's Toolkit is released as Shareware. This is copyrighted material; it is NOT "free software". You are permitted to experiment with this package long enough to determine if it suits your needs, but if you will be making use of the material in your own programs, then a license fee of $50 is required. NO PROGRAM WHICH MAKES USE OF THE MATERIALS IN THIS TOOLKIT MAY BE SOLD COMMERCIALLY, ON A CONTRACT BASIS, OR AS SHAREWARE UNLESS THE SELLER HAS PAID THE LICENSE FEE. Please make all checks and/or money orders payable to: Farpoint Software 2501 Afton Court League City, Texas 77573-3438 If you live in a country other than the United States of America, it will be necessary for you to either (1) Use a check drawn on a U.S. bank specified in U.S. funds, or (2) Use an "International Money Order", available from most banks, once again specified in U.S. funds. When we receive a small check from a non-U.S. bank, both banks charge us a "currency conversion fee" that totals literally almost as much as the check itself. For your convenience and our records, a registration/order form is included in the file ORDER.FRM. As a registered user, you will automatically receive software updates as they become available. You will also receive the source code to the VDFE editor and several experimental voice signal processing and compression programs. Registered users, of course, are given priority if programming assistance or hardware construction assistance is requested. You are granted permission to distribute copies of the Digitized Voice Programmer's Toolkit, provided that (1) no fee is charged for such copies, other that a nominal disk duplication fee, (2) these files are distributed in their original, unmodified form, and (3) ALL the files in the original archive are included with each copy. (See "List of Files" below.) If you paid a "disk duplication fee" or other such fee to a distributor of public domain and shareware programs, be aware that the payment of this fee DOES NOT constitute registration of this Toolkit. Likewise, the payment of a fee to any Bulletin Board Service for the time required to download this Toolkit DOES NOT constitute registration. Registration occurs only through direct interaction with Farpoint Software. If more information is needed, write or contact Alan D. Jones through Compuserve Information Service at user ID 74030,554. To the people who disseminate and distribute shareware: Do NOT strip out the schematic files and distribute this kit without them. This is exactly what happened with the last version, causing a lot of confusion and headaches for both myself and potential users. Do NOT split this archive into more than one file and distribute it in that way, making it almost inevitable that one of the files will get lost. List of Files ------------- The files included with the Digitized Voice Programmer's Toolkit are: READ.ME -- introductory note VOICEKIT.DOC -- this file BUILDIT.DOC -- digitizer construction details and parts list VRMOD.DOC -- descriptions of procedure calls in VRMOD.ASM VPMOD.DOC -- descriptions of procedure calls in VPMOD.ASM ORDER.FRM -- a printable registration form and order form PRINTALL.BAT -- prints out all the above files ADJUST.EXE -- tool for adjusting and testing the digitizer VDFE.EXE -- voice data file editor VDFE.HLP -- help file for the voice data file editor VRMOD.ASM -- assembly routines providing recording capability VRMOD.OBJ -- assembled VRMOD.ASM VPMOD.ASM -- assembly routines providing playback capability VPMOD.OBJ -- assembled VPMOD.ASM VRMOD.H -- include file with C prototypes for VRMOD calls VPMOD.H -- include file with C prototypes for VPMOD calls VRTEST.C -- simple C program demonstrating how to record VRTEST -- make file for VRTEST.C VRTEST.EXE -- executable recording program VPTEST.C -- simple C program demonstrating playback VPTEST -- make file for VPTEST.C VPTEST.EXE -- executable playback program EMBEDDED.C -- simple C program demonstrating embedded voice EMBEDDED -- make file for EMBEDDED.C EMB_VOI.VOI -- voice data for EMBEDDED.C EMB_VOI.PRE -- prefix file used in assembly file creation EMB_VOI.SUF -- suffix file used in assembly file creation EMBEDDED.EXE -- executable with embedded voice data BIN2ASM.EXE -- binary to DB statement converter ULAW.C -- Mu-Law compressor/encoder ULAW -- make file for ULAW.C ULAW.EXE -- executable Mu-Law compressor/encoder UNULAW.C -- Mu-Law decoder UNULAW -- make file for UNULAW.C UNULAW.EXE -- executable Mu-Law decoder FSVPDD.SYS -- voice playback device driver VPDDTEST.C -- simple C program demonstrating use of FSVPDD.SYS VPDDTEST -- make file for VPDDTEST.C VPDDTEST.EXE -- executable using device driver DEMO.VOI -- just a sample of recorded voice RUN_ME.BAT -- plays the demo POSTSCRP.SCH -- schematic for Postscript printer HPGL.SCH -- schematic for HPGL plotter EPSONFX.SCH -- schematic for Epson FX series printer HPLASER.SCH -- schematic for Hewlett-Packard LaserJet printer IBMPRO.SCH -- schematic for IBM Proprinter If you received the Toolkit with any of the above files missing, please notify Farpoint Software. Description of Voice Subroutine Modules --------------------------------------- The key software elements in the kit are two assembly language programs, VRMOD.ASM and VPMOD.ASM, and their assembled OBJ files. These are not stand- alone programs. They are designed to be linked with other programs to provide the voice control routines. The calls associated with recording are in VRMOD, and the calls associated with playback are in VPMOD. Any given program may be linked with either or both of these modules. Typically, a program designed for general distribution would be linked only with VPMOD, since recording requires the hardware device. The external hooks to the two modules consist of various "public" procedure names. All procedures use the Pascal calling convention, since most high-level language compilers can support this calling method. The Pascal calling convention has the following meaning: (1) Procedure names are all caps, and are not preceeded by an underscore. (2) Procedures are called with "far" (intersegment) calls. (3) Short return values appear in the AX register; long return values appear in DX:AX. (4) Parameters are pushed onto the stack in left-to-right order; i.e. the first parameter in the list is pushed first. If the parameter is a doubleword, then the high order word is pushed first. (5) The called subroutine is responsible for clearing the parameters from the stack upon return. The above list will be of interest primarily to assembly language programmers. When working in a high-level language, it is necessary only to make sure that the compiler is using the proper calling method. For C programs, two header files have been included. They are VRMOD.H and VPMOD.H. At the beginning of any C program that uses the voice playback routines, insert the line: #include "vpmod.h" This file contains prototypes of all procedure calls in VPMOD.ASM, declared in a way that causes the compiler to generate correct calling code. The details of how each individual procedure call operates will be found in the separate documents VRMOD.DOC and VPMOD.DOC. It is suggested that you print these files for use as reference material while writing programs. Example Programs ---------------- Note: "Make" files acceptable to Microsoft's NMAKE utility are included for all the example programs. The compiler used was the Microsoft C Compiler version 6.00A. The assembler was the Microsoft Macro Assembler version 5.10. The make files are written to assume that the compiler is installed to include the Large model library and that the default operating system is "neither". If the compiler defaults to DOS, then change the make files so that all occurrences of "llibcer" become "llibce". VRTEST.C (VRTEST.EXE): [Related files: VRTEST, VRMOD.H, VRMOD.ASM] This program is a simple recorder, written to demonstrate the use of procedure calls into VRMOD. To execute the program, first attach the voice digitizer to a COM port and turn the device on; then at the DOS prompt type: VRTEST <port #> <filename> For example, to record from COM2 into file TESTFILE.VOI, type: VRTEST 1 TESTFILE.VOI Recording will begin at the beep. Before recording starts, the program will display a message indicating the size in both bytes and seconds of the available recording buffer. Recording will continue until either a key is pressed or the buffer is full, at which time the buffer will be written to disk. VPTEST.C (VPTEST.EXE): [Related files: VPTEST, VPMOD.H, VPMOD.ASM] This is the counterpart to VRTEST. It demonstrates the use of procedure calls into VPMOD. To execute it, type: VPTEST <filename> The specified file will be loaded into a memory buffer, then played through the speaker. In both this example and VRTEST, messages will be displayed as necessary to indicate the meaning of the value returned by the calibration procedure. FSVPDD.SYS: This is a true DOS device driver which implements the voice playback functions. It calibrates upon initial load, thus removing that responsibility from the calling program. The driver occupies about 3.6k of memory. To load it, place the following line into the CONFIG.SYS file on the boot drive, assuming the driver to be in the VOICE subdirectory on drive C: DEVICE=C:\VOICE\FSVPDD.SYS The driver can then be used by ANY DOS program written in ANY language in which files can be opened, written to, and closed. A mechanism is even provided for playing buffers larger than 64k in size. To use the driver, your program should perform the following actions: (1) Open the file FSVPDD$$ and obtain a file handle or number. (2) Allocate a buffer large enough to hold the voice data plus 4 bytes. (3) Create a 32-bit integer indicating the voice data length in bytes. (4) Copy this integer into the first 4 bytes of the allocated buffer. (5) Fill the remainder of the buffer with the voice data. (6) Perform a FILE WRITE operation to the handle obtained in step 1, giving, as the data source, a pointer to the first byte in the buffer. IMPORTANT: Pass the value 1 as the number of bytes to be written. The driver will take the actual number of bytes from the first 4 bytes in the buffer. (7) (Optional.) Perform a FILE READ operation, again specifying 1 byte, and giving, as the destination, the address of a variable which can hold a 32-bit integer. After the read, this integer will contain the actual number of bytes played. (8) Free the buffer allocated in step 2. (9) Close the file handle. VPDDTEST.C (VPDDTEST.EXE): This is an example written in C which uses the device driver FSVPDD.SYS. It is the smallest of all the example executables, since the device driver handles calibration and playback. The assembly routines do not need to be linked to a program which uses the device driver. EMBEDDED.C (EMBEDDED.EXE): [Related files: EMBEDDED, VPMOD.H, VPMOD.ASM, EMB_VOI.VOI, EMB_VOI.PRE, EMB_VOI.SUF] This is a simple example of the techniques used to embed voice data in an executable program. Instead of reading a separate voice file, the voice data is part of the EXE file. Note that the "make" file in this case is as important to study as the C program. The trick here is to convert the raw binary voice data file into an OBJ file that can be fed through the linker. This is done in three stages: (1) The file-cruncher program BIN2ASM is used to create a file containing only a long list of assembly language DB statements equivalent to the binary data; (2) The prefix file EMB_VOI.PRE and the suffix file EMB_VOI.SUF are combined with the DB statements to form an assembly language module containing all necessary segment brackets and public declarations; (3) This module is assembled and linked with the main program. The contents of the prefix and suffix files depend on the specific application; in this example we use only a single segment and a single block of voice data. A more complex program may contain several modules of this type or have an assortment of labels within a single module. Since the assembler requires segments to be 64k or less, BIN2ASM places a marker comment (a semicolon and a string of minus signs) at each 64k boundary in its output file. If this happens, you must edit the file to end a segment and begin a new one at each of these boundaries. Other Programs -------------- ADJUST.EXE: The purpose of this program is to facilitate the adjustment and testing of digitizer boards. It has no command line parameters; simply type ADJUST to start the program. There are three items of interest on the display: (1) The COM port to which the digitizer is presumably attached while this program runs. Cycle through the ports numbers by pressing <F2> until the correct number appears. (2) A DC offset meter which indicates the value of the average reading from thr A/D converter relative to the "ideal" center point of 80 hex. This reading should be adjusted to zero using R24. (3) A signal level meter. This shows the instantaneous peak-to-peak level being received by the A/D converter. It should be as low as you can get it (while not speaking, of course). A reading of 10 to 20 or less is considered reasonable. If the reading is higher, then possible trouble sources might be: (a) high levels of room background noise from things such as fans and air conditioners, (b) powering the digitizer with an electrically noisy power supply, (c) poor wiring practices on the digitizer or missing bypass capacitors, (d) input gain control R8 set too high, or (e) the digitizer may be sitting too near a source of radiated electrical noise, such as a switching power supply. ULAW.EXE: This program reduces the size of a voice data file by a 2:1 ratio by encoding each 8-bit sample as a 4-bit number such that the quantization error is minimized near the equilibrium (silence) point (80 hex) and increases with increasing excursion. This is a common practice in telephone systems used to reduce data throughput requirements. It causes surprisingly little degradation of the sound quality, although there is some. The effect of quantizing with this program has been described as a slightly "gritty" quality in the reproduced speech. To execute, type: ULAW <source filename> <destination filename> UNULAW.EXE: This program is the matching decoder for ULAW described above. Voice data files encoded with ULAW cannot be played directly; they must first be expanded with UNULAW. The purpose of these two programs is primarily to reduce the size of voice data files for modem transfers. To execute, type: UNULAW <source filename> <destination filename> Source code has been included for both ULAW and UNULAW. BIN2ASM.EXE: This is a program which converts any binary file into a series of DB statements which can be inserted into an assembly language program and thereby made into an object module. Intended to be used in creating programs with internally embedded voice data, its use is demonstrated in EMBEDDED.C. The Voice Data File Editor (VDFE) --------------------------------- This program provides a convenient environment for creating, editing, and generally patching together voice data files, and creating index files for them. VDFE requires no command line parameters. Upon execution, it displays its primary screen and waits for user input. This consists of an assortment of single keystroke commands, accessible directly or through a keyboard operated pulldown menu system. The editor and its operation are described in detail in the file VDFE.HLP; this file can be displayed by pressing the <F1> key while executing VDFE. Graphical Print Files --------------------- These files all contain a copy of the schematic of the digitizer. Several popular printers are supported. The files are: Filename Printer/Plotter -------- --------------- POSTSCRP.SCH Postscript printer HPGL.SCH HPGL plotter EPSONFX.SCH Epson FX series printer HPLASER.SCH Hewlett-Packard LaserJet printer IBMPRO.SCH IBM Proprinter There are some fairly small text strings in the schematic, so all printers are used at their highest resolution. To print or plot the schematic, use the DOS copy command: COPY /B <filename> <output port> For example, to print to a LaserJet connected to LPT1: COPY /B HPLASER.SCH LPT1 General Digitizer Information ----------------------------- Every effort has been made to avoid the use of unusual or hard-to-get parts in the design of this digitizer. For those who prefer not to build the digitizer circuit themselves, it has been made available as a fully built and tested board. The "unpopulated" (etched and drilled but no parts) board is also available. See the price list near the end of this document for details. Unlike the digitizer used in DVPT version 1, which digitized its input directly into the bitstream to be used during playback, this digitizer is a classic 8-bit successive approximation analog-to-digital converter, combined with a microphone amplifier, anti-aliasing filter, level indicators, sample-and-hold amplifier, and a slightly odd 4-bit wide parallel interface to the PC serial port. Other features include a circuit which produces a small degree of amplitude compression to improve perceived loudness, an input gain control with a 20 decibel range (100:1), a DC offset trim, and a remote "stop" switch. The data read from the digitizer is stored as a simple list of 8-bit numbers, each indicating the voltage at the time of the sample. This storage scheme is commonly called PCM or Pulse Code Modulation. The interface to the digitizer depends entirely on direct I/O bit manipulation of the PC serial port, and does NOT use the receiver serial interrupt. It should therefore work equally well connected to any PC serial port, from COM1 through COM4. The interconnecting cable must at least connect pins 3, 4, 5, 6, 7, 8, 20, and 22. When in doubt, use a full 25-wire "straight through" cable; a 2-3-7 or "three-wire" cable will not work. Do not use a null modem cable. The circuit is designed to use a "dynamic" or moving magnetic coil type of microphone. Ceramic or piezoelectric (crystal) microphones can also be made to work by reducing the setting of the input gain control, but these types are often inferior to dynamic microphones. Electret and capacitor microphones won't work unless they provide their owm DC biasing. The type of handheld microphone that often comes with cheap portable cassette tape recorders is a good bet. The power supply must be a split supply; i.e. a positive voltage and a negative voltage (relative to ground) of equal magnitudes. The easiest way to accomplish this is to use two 9 volt batteries. The supply must be at least (+) and (-) 7.5 volts for the circuit to work. A (+) and (-) 12 volt supply can be used as well; however, DON'T use a (+) and (-) 15 volt supply or 15 volt batteries, since this is equal to the absolute maximum rating of the MC1488 line driver and can easily destroy it. Also, DON'T get the supply polarity backwards; if this happens, nearly every semiconductor device on the board will be destroyed. The Signal Level Indicators on the Digitizer -------------------------------------------- The LED labelled D8 is the -6 decibel threshold indicator; it lights when the signal exceeds one half of the maximum range of the A/D converter. The LED labelled D7 is the clipping indicator; it lights when the signal level equals or exceeds the maximum range of the A/D converter. The lighting of the clipping indicator means that the peaks of the reproduced waveform will be clipped or "flat-topped". A small amount of clipping on speech waveforms generally does not degrade intelligibility, although it can be psychologically irritating if it occurs over a significant portion of the data. The level indicators are designed to be used in the following way: When speaking into the microphone or making a recording from another source, the input level (gain) control should be adjusted so that the -6db indicator remains lit a large percentage of the time, whereas the clipping indicator only flickers on the loudest peaks of the sound. This is the optimum signal level. It makes the maximum possible use of the resolution of the A/D converter without introducing an undue amount of clipping distortion. Digitizer Schematic Specifics ----------------------------- This section describes the signal flow through the digitizer. It's probably a good idea to have a printout of the schematic on hand while reading it. U1:A and U1:D together form a microphone amplifier which provides most of the gain of the circuit. Capacitors C3 and C4 give the amplifier a gradual low frequency roll-off, experimentally and subjectively determined to maximize the intelligibility of the final result. The first stage (U1:A) gain can be adjusted with trimpot R8 from a minimum of 1 to a maximum of 100 (actually 101, given that the resistors were that accurate). The second stage (U1:D) provides an additional fixed gain of 23, for a total maximum voltage gain of 2300 for both stages. U1:C, U1:B, and their associated passive components form a fourth order lowpass filter whose cutoff frequency is approximately 7000 Hz. Since the software drives the circuit at a conversion (sampling) rate of about 16500 Hz, the Nyquist criterion requires that frequencies above half of this (8250 Hz) be absent from the analog signal, or aliasing will occur. The lowpass filter approaches this situation well enough to eliminate audible aliasing effects. A full discussion of aliasing would take quite a bit of space and will not be presented here; there are numerous textbooks that cover the subject. R10, R15, D3, D4, D5, and D6 are a crude amplitude (volume) compression circuit. The net effect of the circuit is a mild odd-harmonic distortion of the waveform which we accept in order to improve its dynamic range. The circuit causes the peak-to-peak gain of high-level signals to be less than that of low-level signals. This allows us to operate at higher overall input levels before severe clipping occurs, causing somewhat improved resolution (reduced quantization error) in the lower-volume sounds that contribute so much to the perceived "clarity" of a reproduced speech signal. There is also an increase in the perceived overall volume at the output. C5 is a DC blocking capacitor. The appropriate DC offset required by the analog-to-digital converter is supplied by R9 and R16 or trimpot R24. D1 and D2 protect the circuitry which follows from overvoltages. U5 is a sample-and- hold amplifier, whose purpose is to maintain the input signal to the converter at a steady voltage during the actual conversion. U6 is an 8-bit successive- approximation analog-to-digital (A/D) converter capable of performing one conversion every 40 microseconds. Its conversion rate limit is therefore 25 KHz, more than sufficient for this application. The 8-bit result is multiplexed onto 4 lines by either U4 or U7, and the logic levels are changed to RS-232 levels by U3. U8 is an RS-232 receiver used to drive the control inputs on U5, U6, U4, and U7. Dual opamp U2 is used as a visual signal level indicator. The opamps are wired as comparators. When the excursion of the signal being fed to U5 goes below ground, U2:A triggers, lighting LED D7 to indicate clipping. Note that the input signal range of the A/D converter is set up to range from zero volts to about 4 volts. The input signal needs to be only at one-half of the clipping amplitude in order to trigger U2:B, thus lighting LED D8. One-half amplitude equals 6 decibels below clipping. U9 is a three terminal linear 5 volt regulator. It provides power to the ICs which must operate from a regulated 5 volt logic supply (U6, U4, U7, U8). All the capacitors shown in parallel groups in the lower left area of the schematic are power supply bypass capacitors. Their purpose is to provide each IC with isolation from the noise induced on the power supply rails by the other parts of the circuit. These capacitors should be distributed around the board so that each power supply pin of each IC is connected directly to a bypass capacitor or a pair of bypass capacitors whose other terminal is solidly connected to ground. Components U1, U2, and U5 should have two bypass capacitors per power supply pin: one 0.1 uF and one 22 uF tantalum. There are two locations in the schematic where redundant components are shown in order to increase versatility or availability: (1) Use EITHER R9 and R16 OR trimpot R24, but not both; R24 is recommended. (2) Use EITHER U4 OR U7, but not both; selection depends on availability. See the file BUILDIT.DOC for more construction details and a parts list. Playing Back Through the PC Speaker ----------------------------------- The speaker on the PC and its associated driver circuitry is quite simple and crude, having been designed primarily for creating single square-wave tones of various audio frequencies. This speaker is typically driven by a pair of transistors used as a current amplifier which is in turn driven directly by the output of a TTL gate. This results in only two possibilities of voltage across the voice coil: 0 volts and 5 volts. Any sound to be reproduced by this system must be reduced to an approximation in the form of a stream of constant-amplitude rectangular pulses of varying duration and frequency. The exact details of the playback mechanism used in this package is probably best studied by looking at the code in VPMOD.ASM. Suffice it to say here that instantaneous DC drive voltages to the speaker are simulated by providing an alternating series of high and low output pulses such that the ratio of high pulses to low pulses averages out to the desired voltage. The lowpass filtering required is provided by the human ear and the mechanical limits of the speaker. Because of the nature of the speaker drive, the transition timings are quantized into discrete equally-spaced time intervals. The software has been written to minimize this interval (one NOP instruction) and therefore maximize the apparent "DC" resolution. The return value from the PCALIBRATE routine contains a "delay counter" value. This value is equal to the number of discrete steps available; the faster the computer, the greater the number of steps. Any number greater than or equal to 256 produces optimum sound reproduction, since this is the resolution at which the sound was originally recorded. As the number of discrete steps decreases below 256, the sound quality degrades steadily, becoming moderately bad at 64. It is at this point that the PCALIBRATE routine declares the computer to be too slow. This number is reached when the CPU becomes an 8 MHz 286, so your programs should be targeted at systems with at least this level of performance. And yes, if you have already looked, there is indeed self-modifying code in VPMOD.ASM. Whatever it takes to get the resolution. Prices for Software and Hardware -------------------------------- Registration for Software ------------- $50 Becoming registered gives you a license to use portions of the code included with this toolkit in your own programs. It does not grant a license to sell any of the complete working programs in this toolkit intact as part of another package, with the sole exception of the device driver, FSVPDD.SYS. If sold intact as part of another package, this device driver may NOT be altered in any way. - - - - - - - - - - - - - - - - - - - - - - - - - - Digitizer: UNREGISTERED users REGISTERED users ------------------ ---------------- Fully complete and tested digitizer board with 30 day warranty (does not include enclosure, batteries, serial cable, or microphone) ---------------------- $79 ----------- $59 Unpopulated digitizer printed circuit board only, etched and silkscreened, without parts ----------------------------- $21 ----------- $17 Other items: Schematic paper print ---------------------- $5 ----------- Free - - - - - - - - - - - - - - - - - - - - - - - - - - Incidental hardware: Although we can supply the following items, we recommend that you obtain them through your local consumer electronics stores and computer stores. Microphone --------------------------- $19 DB-25 cable -------------------------- $20 DB-9 to DB-25 adapter ---------------- $10 - - - - - - - - - - - - - - - - - - - - - - - - - - NOTICE: Since our manufacturing costs may increase as parts costs increase over time, the prices quoted for hardware items are valid only through December 1992. Please address all correspondence to: Farpoint Software 2501 Afton Court League City, Texas 77573-3438 FAX: (713) 332-4730 Compuserve ID: 74030,554 If at all possible, use the form created by printing the file ORDER.FRM. If you choose to write in the information by hand, please print clearly. We have received several registration forms for version 1 which were quite difficult to read.