home
***
CD-ROM
|
disk
|
FTP
|
other
***
search
/
HPAVC
/
HPAVC CD-ROM.iso
/
CIS_GAME.ARJ
/
QSBSB1.THD
< prev
next >
Wrap
Text File
|
1993-07-01
|
35KB
|
721 lines
________________________ Subj: Playing Sound on PC ________________________
Fm: Mark Betz/GD SL 76605,2346 # 181875
To: Intergalactic, Dev., Inc 76356,2172 (X) Date: 27-Jun-92 13:08:37
A couple of choices, Ezra: You can roll your own. This will involve creating
a low-level driver (probably driven off the timer interrupt). Or, you can
purchase a library, such as John Ratcliff's DIGPAK, which supports a variety
of sound devices. Adlib .ROL, and CMS .CMF files are essentially MIDI files
with restricted MIDI event/message recognition. The files are parsed, and the
driver will turn notes on and off at the proper times. I've done low-level
programming to the FM chip, but haven't had occasion to parse MIDI files yet.
I imagine it's not trivial programming, as the timing must be fairly precise.
--Mark
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 181904
To: Sarwan Narine 76675,164 Date: 27-Jun-92 14:52:53
Hi, Sarwan. There are several routes you can go. The DSP chip on the
Soundblaster (and TB) can be driven via DMA, freeing the CPU for other
business. Or it can be "pumped" using the timer to send data to it at the
correct speed. Whichever route you take will be fairly complex and require
low-level programming to the sound card hardware. A third alternative is to
use the supplied drivers, which can be loaded and accessed by your program,
and a fourth is to purchase a sound devide library like DIGPAK. For
alternatives 1, 2, and 3, get yourself a copy of the Sound Blaster
Developer's Kit from CMS. It's not great, but you can use it to get the job
done, and there is good information on the low-level interfaces to the
hardware.
Creative Labs, Inc.
2050 Duane Ave.
Santa Clara, CA 95054
Sorry! I don't have the phone number at hand.
--Mark
...........................................................................
Fm: Rasch H Young 70511,2043 # 182020
To: Sarwan Narine 76675,164 Date: 27-Jun-92 22:24:51
Sarwan,
I have both the Sound blaster SDK and DIgPak. I believe DIGPAK pro is a
better product at a lower price $69.95. The Audio Solution 1-314-567-0267.
Rasch
________________________ Subj: Sound card detection ________________________
Fm: Sarwan Narine 76675,164 # 185506
To: All Date: 08-Jul-92 02:47:07
How do you detect the presence of a sound card in a PC system?
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 187160
To: Sarwan Narine 76675,164 (X) Date: 12-Jul-92 19:54:16
Hi, Sarwan. Here is a function (and support functions) for detecting the
presence of the FM music chip. If DetectFM() returns true then the chip is
present in the system. This tells you that you have either an SB, Adlib, or
one of the SB clones. You can further distinguish between the SB and Adlib,
but more on that later. The first function we need is a function to send data
to the FM chip registers. Here's one in assembler...
_SetReg proc far
push bp
mov bp, sp
push ax
push cx
push dx
xor ax, ax
mov dx, 0388h ; FM Address Register Port
mov al, byte ptr [bp + 6] ; get the register into al
out dx, al ; write it to the Address Reg.
mov cx, 7 ; need to wait 3.3 microsecs here...
wait1: ; which we'll do by reading the
in al, dx ; status reg 7 times
loop wait1
mov al, byte ptr [bp + 8] ; get the data byte into al
inc dx ; point dx to the Data Register Port
out dx, al ; write the data
mov cx, 35 ; need to wait 23 microsecs here...
wait2: ; read the port 35 times
in al, dx
loop wait2
pop dx
pop cx
pop ax
pop bp
retf
_SetReg endp
The techniques for this come right out of the Sblaster Developer's Kit, which
I recommend. It's not the job I wish it was, but it's all that is available
as far as I know. I haven't seen any books on the topic. Anyway, the SBDK is
adequate if you're willing to go mining yourself for the stuff it doesn't
give you. Call CMS at 408-986-1461 if you're interested. Now that we have the
function to set the registers we need the the function that detects the card.
This writes to some registers, which are referred to in the calls to
SetReg(). I won't explain what they do, as this is a topic for a long
magazine article, or chapter in a book.
// DetectFM() returns true if the FM chip is detected in the system, or
// false if it is not.
bool DetectFM() {
byte stat1;
byte stat2;
byte result = 0;
SetReg( 0x1, 0 ); // initialize test register
SetReg( 0x4, 0x60 ); // reset both timers
SetReg( 0x4, 0x80 ); // enable timer interrupts
stat1 = inportb( 0x388 ); // read the status port
SetReg( 0x2, 0xff ); // write ff to the timer 1 count
SetReg( 0x4, 0x21 ); // start timer 1
Wait( 90 ); // wait 90 microseconds
stat2 = inportb( 0x388 ); // read the status port
SetReg( 0x4, 0x60 ); // reset both timers
SetReg( 0x4, 0x80 ); // enable timer interrupts
stat1 &= 0xe0; // stat1 should be zero after
if ( !stat1 ) { // ANDing it with 0xe0
stat2 &= 0xe0; // stat2 should be 0xc0 after
if ( stat2 == 0xc0 ) { // ANDing it with 0xe0
result++; // if so, return true
}
}
return( result );
}
Oh, and I see we have one other support function required. Wait() simply
waits for the specified number of microseconds, approximately. You can
substitute your own function if you like, but here is the technique used in
the SBDK. I would probably implement it as a more accurate timer-based
function, instead of reading the bus the way they do, but I haven't had the
time to re-do it.
// Wait() approximates a wait in microseconds by reading the bus. Called
// only by the DetectFM() function.
void Wait( int howLong )
{
byte dummy;
int reads = howLong / 10;
reads *= 16;
do
{
dummy = inportb( 0x388 );
reads--;
} while ( reads );
return;
}
One last thing. I mentioned that you can tell whether it's a Sound Blaster FM
chip you're detecting. This is because the SB maps the chip to ports 0x2x8
and 0x2x9 (where x is the base port address of the SB). The Adlib standard is
to map the chip to 0x388 and 0x389. To maintain compatibility the SB decodes
the chip here as well. So if you can find the chip at both locations it's a
Sound Blaster. If not it's an Adlib.
________________________ Subj: Sound card I/O ports ________________________
Fm: Sarwan Narine 76675,164 # 187955
To: Mark Betz/GD SL 76605,2346 (X) Date: 15-Jul-92 02:46:26
Mark, thank you for that informative and complete example of how to detect a
sound card.
On my ThunderBoard sound card I can set the I/O port addresses to 210h, 220h,
230h, 240h, 250h, or 260h. How do you find out where the I/O ports are? The
documentation I have says that only ports 2n6, 2n8, 2n9, 2nA, 2nC, and 2nE
are used, where n is 1,2,3,4,5,or 6. What are these ports? A brief
description of each port would be appreciated. Thank you.
--Sarwan-
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 188069
To: Sarwan Narine 76675,164 Date: 15-Jul-92 15:12:08
Hi, Sarwan. According to my reference the SB uses the addresses from 2n0H to
2nFH where n is in the range 1 to 6 inclusive. Here is a map of the ports:
PORT WHAT IS IT WHAT CAN I DO TO IT
------------------------------------------------------------------
200H-207H ANALOG JOYSTICK READ/WRITE
2n0H C/MS MUSIC VOICE 1-6 DATA PORT WRITE ONLY
2n1H C/MS MUSIC VOICE 1-6 REGISTER PORT WRITE ONLY
2n2H C/MS MUSIC VOICE 7-12 DATA PORT WRITE ONLY
2n3H C/MS MUSIC VOICE 7-12 REGISTER PORT WRITE ONLY
2n6H DSP RESET WRITE ONLY
2n8H FM MUSIC STATUS PORT READ
2n8H FM MUSIC REGISTER PORT WRITE
2n9H FM MUSIC DATA PORT WRITE ONLY
2nAH DSP READ DATA PORT READ ONLY
2nCH DSP WRITE DATA OR COMMAND WRITE
2nCH DSP WRITE BUFFER STATUS (BIT 7) READ
2nEH DSP DATA AVAILABLE STATUS (BIT 7) READ ONLY
The C/MS music voice ports are available only on SB cards with these chips
installed. The FM music ports can also be accessed at ports 388H and 389H
(for Adlib compatibility).
--Mark
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 189758
To: Sarwan Narine 76675,164 Date: 20-Jul-92 16:21:03
>> ...how do you find the value of n
Well, there are a couple of options. You can ask the user, which is the route
most apps take. You can rely on Creative's SBLASTER environment variable
(NOT!). Or you can perform the test I outlined for each possible port.
--Mark
_________________________ Subj: Playing .VOC files _________________________
Fm: Sarwan Narine 76675,164 # 191206
To: All Date: 24-Jul-92 04:12:35
How do you output an 8-bit sample of a .VOC file through the SoundBlaster or
AdLib sound card? Also, is there any on-line information on using a DMA
channel or timer to playback sound? I'm looking for assembly source code.
Thank you.
--Sarwan-
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 191420
To: Sarwan Narine 76675,164 Date: 24-Jul-92 20:02:33
Hello again, Sarwan. If you have the manual for the Sound Blaster you will
find a simple reference to the functions in the CT-VOICE.DRV driver supplied
on disk with the card. This driver is the simplest way to pump data to the
dsp chip, since it provides you with some high level control constructs.
Alternatively you can write directly to the chip. The DSP chip is actually
very simple, being sort of like a pipe with one end attached to memory, and
the other to the speaker system. All you really have to do is pour data into
it at the proper rate. You can do this either by pumping it through with the
timer interrupt, or using the DMA channel 1. I strongly suggest contacting
Creative Labs and buying the SB developer's kit.
If you simply want to get sound support, then I suggest the DIGPAK library.
Leave mail or a message here for John Ratcliff 70253,3237. The have a lib for
sale at a very good price which supports midi and digitized sound on just
about every device you can imagine, if half of what I've heard is true.
--Mark
...........................................................................
Fm: James Mayes 71151,3037 # 203245
To: Mark Betz/GD SL 76605,2346 (X) Date: 21-Aug-92 18:04:08
I also have another question. I recall your recently discussing with
someone else your experience programming the Soundblaster by loading the
driver into memory and then making calls to the driver with the correct
registers initialized properly. I have always used the SB Development Kit
by Creative, but I am curious about the direct method and I was wondering
about a few things. (BTW, I am strictly talking about digitized voice and
the CT-VOICE driver)
1. The SBK says you need to load the driver at offset 0 of some
segment, load the registers and make a far CALL to that seg,off. What
would be the syntax for making this call from C given a far pointer to
the buffer containing the driver?
2. The CT-VOICE driver doesn't include any functions for detecting the
SB card or scanning for the correct interrupt number. Are you aware of
any good sources for this info?
3. Do you know if there would be a problem with distributing the
CT-VOICE driver file with a game that used this approach? (I don't
think there should be, but those kind of questions always worry me.)
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 203304
To: James Mayes 71151,3037 (X) Date: 21-Aug-92 19:47:00
Hi, James. I'm not sure what you mean by "the direct method". Is there some
other way to access CT-VOICE.DRV explained in the SDK? Basically you have to
get the driver into memory first. That requires allocating memory for it, and
reading it in. The twist is that you have to allocate an extra paragraph, or
16 bytes, so that you can force the offset to 0. the new keyword and alloc()
functions usually return an offset of 4, and in any case you can't guarantee
that the offset will be 0. Allocate an extra paragraph, and do a little
segment/offset arithmetic to normalize the pointer (make sure to keep a copy
of the original for use with free() or delete).
Once you get the driver read into memory you can set the IRQ and port address
directly. The offsets of these values are listed in the SDK, but anyway, they
are right near the beginning of the CT-VOICE driver file; maybe offset 40h or
thereabouts. As for finding out which values to use, you either have to ask
the user, or test all possible combinations.
From that point on it's a simple matter of casting the driver buffer pointer
to a pointer to function, and then making calls to it.
void* buf; // pointer to buffer void (*drvr)();
// function pointer buf = new byte[sizeof_driver]; // alloc driver
memory drvr = (void(*)()) buf; // not sure about this cast!
...........................................................................
Fm: Sarwan Narine 76675,164 # 203407
To: All Date: 22-Aug-92 02:06:32
Well, I've posted this question before, with unsucessful results. So, I'll
post it again -- because I like to aggravate people <g>.
OK, here it is: How do you output an 8-bit sample to the SoundBlaster card?
Assume the SB is installed at 220h. Thanks.
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 203510
To: Sarwan Narine 76675,164 (X) Date: 22-Aug-92 13:39:16
Sarwan, if you're talking about driving the output from your application,
then you are into several complex areas of hardware programming, including
the interrupt controller, dma controller, and the dsp chip on the SB. I'll
give you skeletal procedures for doing it with the driver, but you'll have to
dig in your references for the details.
Using the driver: 1. Allocate ram for the driver on the heap. The driver
needs to be loaded at offset 0 of a segment, so grab
a paragraph more than you need, and do pointer math
to force the offset to 0 and the seg to the largest
possible value.
2. Read the driver (CT-VOICE.DRV) into memory.
3. Set a function pointer on the first byte of the driver
NOTE: to call a driver function set BX = the function no.,
and set any other registers needed, then call the driver.
4. Set the base I/O address. BX = 1, AX = base I/O addr.
5. Set the dma interrupt. BX = 2, AX = dma interrupt no.
6. Initialize the driver. BX = 3. Returns error code, AX.
AX = 0: driver init successful
AX = 1: bad driver version
AX = 2: I/O read/write failure
AX = 3: DMA interrupt failure
7. Load the voice data into memory (see block structure)
8. Turn the speaker on. BX = 4, AL != 0.
9. Start voice output. BX = 6, ES:DI = voice data, returns
error code, AX.
AX = 0: output successful
AX != 0: failure
10. Turn speaker off. BX = 4, AL = 0.
11. Terminate driver. BX = 9.
Note that the voice data you output must be in CMS .VOC format. This format
uses a block structure, with each block of data preceeded by a header which
identifies it to the driver. The only block-type you're concerned with at
this point is the Voice Data Block. It looks like this:
byte 0 : type field = 01
1-3: three bytes data block length (actual length - 2)
4 : time constant = 256- (1000000 / sampling rate)
5 : compress. (0: raw, 1: 4-bit, 2: 2.6-bit, 3: 2-bit)
Make sure one of these blocks is at the front of your voice data and
initialized correctly. Once you start the voice output you'll get control of
the processor back, and your program can continue.
...........................................................................
Fm: James Mayes 71151,3037 # 203802
To: Mark Betz/GD SL 76605,2346 (X) Date: 23-Aug-92 10:45:17
Thanks a lot, Mark. By "direct method" I simply meant writing my own code
to access the driver rather than calling the ctvm_XXXX routines supplied
with the SDK. The longer I program, the worse my English becomes <g>. The
info you gave me was exactly what I was looking for. I think I will give it
all a try very soon. Thanks for your help!!
...........................................................................
Fm: Sarwan Narine 76675,164 # 207584
To: Mark Betz 76605,2346 (X) Date: 02-Sep-92 02:51:10
Hi Mark. I haven't had much success with playing back a .VOC file thru the
SoundBlaster. Here's what I've got (assume SB installed at 220h):
; reset DSP
OUT 226h, 1
OUT 226h, 0
; turn on the speaker
OUT 22Ch, 0D1h
; playback loop
PLAYBACK_LOOP: OUT 22Ch, 10h
OUT 22Ch, SAMPLE
DELAY
LOOP PLAYBACK_LOOP
; turn off the speaker
OUT 22Ch, 0D3h
Hey kids, don't try this at home. The above is "pseudo-code", don't try it
on your assembler <g>.
OK, Mark, the above _occasionally_ outputs something resembling the .VOC file
contents, although it sounds like it's played-back in slow-motion. Most of
the times I get no output. Any suggestions?
...........................................................................
Fm: Mark Betz/GD SL 76605,2346 # 207964
To: Sarwan Narine 76675,164 (X) Date: 02-Sep-92 23:34:56
Hi, Sarwan. The procedure outlined in the manual says that you must wait 3
microseconds between the two steps of the DSP reset process. It could be that
this is mandatory, and it could be intermittent because certain variances
caused by a cache hit/non-hit or something are causing you to get a >= 3
microsecond delay on some calls, and not on others. I'd specifically place
the delay in the function. As for slow playback, that is almost certainly
related to the speed with which you're sending the data to the DSP.
...........................................................................
Fm: Hans Peter Rushworth 100031,473 # 214082
To: Mark Betz/GD SL 76605,2346 (X) Date: 17-Sep-92 10:08:04
>> The volume can be controlled by altering the sampled data in certain
mathmatically precise ways that I do not know <g>.
It is much simpler than you think!
If the samples are simple signed amplitudes, then just multiply each 8-bit
sample by a constant from 0 to 256, and use the top 8 bits of the result. If
they are unsigned, convert to signed by subtracting 128 first, and adding 128
afterwards. (you must use a signed multiply or equivalent shift operation).
The output volume is not linearly related to the multiplier. A range of 0 1 2
4 8 16 32 ... (ie use shifts!) will give a more or less linear progressive
increase in volume (ie half the amplitude is NOT half as loud). With low
multipliers the quantisation error (hence noise) will be high, unfortunately.
Only 8-bits are a bit limiting I'm afraid (pun intended), so for a finer
variation in volume you may need find intermediate values for the multiplier.
For example, the multiplier between 128 and 256 is 181, (not 192) which is
found by taking 2^7.5 (2^7 is 128 and 2^8 is 256, so 2^7.5 is half way in
between).
You shouldn't simply add values as you suggest, since this will introduce
high levels harmonic distortion, not what you want I suspect.
_______________________ Subj: Sound Blaster Effects _______________________
Fm: Jesse 76646,3302 # 267987
To: all Date: 26-Dec-92 15:32:45
Here's a thought, though it's either easy to do and has already been done, or
it can't be done and hasn't [g]
Instead of worrying about .vocs for small sound effects like gunfire,
explosions, engine noise, etc, couldn't we just create an instrument for each
sound and dynamically "play" them as the game goes on? I've only peeked at
the FM-Organ thing that came with the SB, but I seem to recall it letting
several instruments play at once.
...........................................................................
Fm: John W. Ratcliff 70253,3237 # 268036
To: Jesse 76646,3302 Date: 26-Dec-92 17:43:38
Jesse,
>>just create an instrument for each sound and dynamically "play" them as the
game goes on?
Yes it can be done. That is in fact how all sound effects are made in Ultima
Underworld and a number of other games. However, it is very difficult to
do, has no analgy on non-FM synth devices, and digital sound effects are
easier and sound better. Plus I'm selling DigPak and people should definatly
be doing digital effects from my point of view. You can do all of you
music/FX that way if you use the Audio Interface Library from Miles Design
Inc. However, AIL has a rather overwhelming API and is out of the price
range of the small developer.
John.
...........................................................................
Fm: Mark Betz/Ass't SysOp 76605,2346 # 268883
To: Jesse 76646,3302 (X) Date: 28-Dec-92 11:26:54
Hi, Jesse. I don't know how many people use digitized sound for engine
noises. The engine is one area where a synth chip can do a fine job, so my
approach has been to create the engine noises on the synth, leaving the
sampled sound to be used for specific effects.
_____________________ Subj: Sound Blaster Programming _____________________
Fm: Everett Kaser (Sherlock) 70673,1547 # 310695
To: Vu Truong (Siliconis) 70242,3015 (X) Date: 08-Mar-93 22:10:28
There's a file available from an FTP site on internet (and may be somewhere
here on CIS) written by Jeffrey S. Lee for the Soundblaster Freedom Project.
It's file name is usually SFP.ZIP. It contains lots of good reference
information regarding the Soundblaster (and Adlib) registers. I have a
couple of other files that I've snarfed off of internet that contain file
formats for SBI files and CMF files. If they're not already available in one
of the libraries, and someone would point out the appropriate place for them,
I'd be glad to upload them (on Tuesday, for a hamburger today <G>).
Everett
...........................................................................
Fm: Dan Corritore 70243,1110 # 310714
To: Vu Truong (Siliconis) 70242,3015 (X) Date: 08-Mar-93 22:36:29
I have the format here, but I'd rather not type the long descriptions that
are on it. If you want more description, buy 'The Sound Blaster Book'--very
useful, especially in the file-formats category.
Bytes Description
0-3 File ID-- 'SBI' followed by 0x1a
4-35 Instrument Name--name of instrument (ASCIIZ form)
36 Modulator sound properties
37 Carrier sound properties
38 Modulator volume
39 Carrier volume
40 Modulator attack/delay
41 Carrier attack/delay
42 Modulator sustain/release
43 Carrier sustain/release
44 Modulator waveform
45 Carrier waveform
46 Synthesis mode and phase shifting
47-51 Future use
Sorry if the above does not appear in columns.. my software doesn't give a
'WYSIWYG' interface in messages. And also sorry that the above weren't very
descriptive!
_Dan
...........................................................................
Fm: Everett Kaser (Sherlock) 70673,1547 # 313151
To: Everett Kaser (Sherlock) 70673,1547 Date: 13-Mar-93 15:07:10
OK, here's the internet ftp sites for soundboard related stuff that I had at
work and their directories:
ftp.brad.ac.uk misc/mod
saffron.inset.com pub/unix/sound_drivers (access limited to 3:30pm-5:30am)
ftp.ulowell.edu msdos/Sound/AdlibSB
wuarchive.wustl.edu mirrors3/garbo.uwasa.fi/sb
snake.mcs.kent.edu pub/SB-Adlib
ccosun.caltech.edu pub/heathh/sb
Good luck! I hope you find what you want.
Everett
...........................................................................
Fm: Everett Kaser (Sherlock) 70673,1547 # 313152
To: Vu Truong (Siliconis) 70242,3015 (X) Date: 13-Mar-93 15:07:30
Ok, I was unable to track down the file itself (that documents the .SBI file
format) in my CAREFULLY arranged and METICULOUSLY indexed disk library <G>,
so I typed in from a printed copy the section pertaining to the SBI format:
-----------------------------------------------------------------------
Sound Blaster Instruments File (SBI) Format:
----------------------------------------------
The SBFM driver programs the registers of the FM chips to synthesize the
different instrument sound. Such register information is stored in the SBI
file with one instrument per file.
The Sound Blaster IEDIT program allows users to define new instruments and
store them in the SBI format.
The format of the SBI file is as follows:
Offset (hex): Description: -------------
------------------------------------------
00 - 03 File ID (ASCII string "SBI", ends with 1A hex)
04 - 23 Instrument Name (NULL terminated ASCII string).
24 Modulator Sound Characteristic
25 Carrier Sound Characteristic
Bit 7 : Pitch Vibrato (AM)
Bit 6 : Amplitude Vibrato (VIB)
Bit 5 : Sustaining Sound (EG-TYP)
Bit 4 : Envelope Scaling (KSR)
Bit 3-0 : Frequency Multiplier (MULTIPLE)
26 Modulator Scaling/Output Level
27 Carrier Scaling/Output Level
Bit 7-6 : Level Scaling (KSL)
Bit 5-0 : Output Level (TL)
28 Modulator Attack/Decay
29 Carrier Attack/Decay
Bit 7-4 : Attack Rate (AR)
Bit 3-0 : Decay Rate (DR)
2A Modulator Sustain Level/Release Rate
2B Carrier Sustain Level/Release Rate
Bit 7-4 : Sustain Level (SL)
Bit 3-0 : Release Rate (RR)
2C Modulator Wave Select
2D Carrier Wave Select
Bit 7-2 : All bits clear
Bit 1-0 : Wave Select (WS)
2E Feedback/Connection
Bit 7-4 : all bits clear
Bit 3-1 : Modulator Feedback (FB)
Bit 0 : Connection
2F - 33 Reserved for future use
------------------------- that's all folks! ------------------------------
Everett
_______________________ Subj: VOC Compression Method _______________________
Fm: Bryant Bunderson 71742,1076 # 378740
To: All Date: 18-Jun-93 15:43:46
I am trying to write a program to compress my PCM sound files to the 4 bit
ADPCM format used in the Sound Blaster VOC files so they don't take up so
they don't take up so much space. I have only been partially successful.
Does anyone out there have a specification or code fragment describing the
compression algorithm used in 4 bit compressed VOC files?
I have seen a lot of technical know how in this forum and thought I would ask
here. Any help would be appreciated.
Thanks, Bryant Bunderson
...........................................................................
Fm: Bryant Bunderson 71742,1076 # 379544
To: Andrew Amess 100141,1567 Date: 19-Jun-93 23:25:44
Yes, the ADPCM method does lose information. It stands for Adaptive Delta
Pulse Code Modulation. Basically instead of saving all the bits of a sample
(PCM) it stores the difference or delta between two samples using fewer bits.
Generally the fewer bits you use to represent the deltas the less the
resulting wave looks like the original.
I need my compressed sound files to be compatible with the Sound Blaster VOC
format so I really should use thier method of ADPCM compression but thanks
for the response.
Bryant
...........................................................................
Fm: Patrick Reilly 71333,2764 # 379632
To: Bryant Bunderson 71742,1076 Date: 20-Jun-93 03:20:04
Bryant,
I though the Sound Blaster used a modified form of ADPCM (I know that's
what *my* code used <g>) - a sentinel value is used to allow the following
word to be full (in this case) 8-bit. For example, using 3-bit encoding, 0-6
are delta values; a 7 indicates that the decoder should grab the following
full 8-bits and use this as an absolute value instead of a delta. If the
system is designed right, you can get nice size savings; I had some 8-bit
sound files that ended up being (average) about the same size using 2-bit;
for 3-bit, they averaged about 2/3, etc. In fact, it varied so widely from
file to file that I ended up using a file header and allowing different bit
sizes. The decode program (read in .WAV files) would go through and try 2, 3,
4, 5, and 6-bit encoding; whoever came out smallest won; if they were all
bigger (never happened) used a special header signature to show that it was a
raw 8-bit data file... Worked pretty well with PC-speaker sound library...
-Pat-
...........................................................................
Fm: Kirk Bateman 100112,71 # 379643
To: Bryant Bunderson 71742,1076 Date: 20-Jun-93 05:57:22
I wouldn't use the ADPCM routine SB style to compress your SB data because
the loss in quality is too great, look in the CLMFORUM under the Data
Compression Section there is some interresting work there.
---Kirk/UK
...........................................................................
Fm: Bryant Bunderson 71742,1076 # 380900
To: Patrick Reilly 71333,2764 (X) Date: 21-Jun-93 18:45:14
I will look for a "sentinal bit" change in the data stream and see if I can
determine if it is being used. I had come to the conclusion that they were
doing something like that but didn't see a pattern. At least not a byte
aligned pattern. Thanks for the help.
Bryant
P.S. Do your internal speaker routines work pretty well? I have pretty much
crashed and burned every time I try to get intellagible playback through the
internal speaker.
...........................................................................
Fm: Patrick Reilly 71333,2764 # 381245
To: Bryant Bunderson 71742,1076 (X) Date: 22-Jun-93 02:15:00
Bryant,
Sentinel value - I know that for *my* encoding/decoding, what I use for the
sentinel value is the 'odd man out'. For example: with 4-bit encoding, this
means that you have 1 sign bit and 3 'value' bits, for a range from +7 to -8.
In this case, -8 is the sentinel value, and only +7..-7 are delta values (-8
is 1000b).
Yeah, my speaker lib worked fine; I only ran it in the foreground though
(actually, it ran in the background, but I used a while-loop to do nothing
until the buffer finished playing). I would seriously think that there would
be some files here on CIS that have a speaker driver...
-Pat-
...........................................................................
Fm: John W. Ratcliff 70253,3237 # 381154
To: Bryant Bunderson 71742,1076 (X) Date: 21-Jun-93 23:30:31
An algorithm called ACOMP is in the public domain. It was published in the
July 1992 issue of DDJ. It uses a special type of ADPCM that features a user
selectable amount of loss, supports silence squelching, and provides
extremely fast decompression speed.
Go DDJFORUM to find the source.
John.
...........................................................................
