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Sound Blaster 16 (view source)

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(Detecting DSP, read/write to the ports, code, playing audio)
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{{In Progress}}
 
The Sound Blaster series is a family of sound cards made by Creative. For many years they were the standard audio cards on IBM PC's.
 
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==Digital Signal Processor==
The DSP built into the Sound Blaster 16 supports playing and recording audio in 8-bit and 16-bit PCM encoded samples, along with playing several other formats (ADPCM, etc.). The base I/O register address can be found using the PCI bus for PCI models, or by detecting the presence of an older ISA Sound Blaster by issuing a Get Version command to one of several common I/O port addresses (0x220, 0x240, etc.) and waiting for a response.
To play audio, the ISA DMA controller must be programmed, then the DSP must be issued a Set Output Sample Rate command, and a Play command that includes the format of the audio (mono/stereo, signed, etc.) and a sample count.
 
Base port for DSP ports is 0x220.
 
{| {{wikitable}}
! OffsetPort
! Description
|-
| 0x224 || DSP Mixer port
| 0x04 || Mixer Port (Set to 0x82 for Interrupt Status)
|-
| 0x050x225 || DSP Mixer Data (Read to get Interruptdata Status)port
|-
| 0x060x226 || DSP Reset
|-
| 0x0a0x22A || DSP Read
|-
| 0x0c0x22C || DSP Write (Read this port for Write Status)
|-
| 0x0e0x22E || DSP Read Status (Read this port to acknowledge 8-bit interrupt)
|-
| 0x0f0x22F || DSP 16-bit Interrupt Acknowledge (Read this port to acknowledge 16-bit interrupt) (DSP Version 4.0+ Only)
|-
|}
 
{| {{wikitable}}
! Command for DSP Write
! Description
! Output/input for command
|-
| 0x410x40 || Set Outputtime Sampleconstant Rate|| 8 bit value
|-
| 0x41 || Set Output Sample Rate || high bit/low bit
| 0xa6 || Play PCM (Auto Initialize DMA, FIFO Enabled)
|-
| 0xd90xD1 || StopTurn PCMspeaker (Autoon Initialize DMA)||
|-
| 0xe10xD3 || GetTurn DSPspeaker Versionoff ||
|-
| 0xD0 || Stop playing 8 bit channel ||
|-
| 0xD4 || Resume playback of 8 bit channel ||
|-
| 0xD5 || Stop playing 16 bit channel ||
|-
| 0xD6 || Resume playback of 16 bit channel ||
|-
| 0xE1 || Get DSP version || major version/minor version
|-
|}
 
{| {{wikitable}}
When the samples have been played, an interrupt will be fired, and the OS/Driver will have the chance to refill the buffer, or to issue a Stop command.
! Command for Mixer port
! Description
! Mixer data port
|-
| 0x22 || Master volume || 0xLR L=left volume R=right volume min=0x0 max=0xF (default value is 0xCC or 0x11)
|-
| 0x80 || Set IRQ || See below
|-
|}
 
There are two modes for transfering data. First is single mode - data from buffer are played, interrupt is fired and playing is stopped. It make lower quality of sound. Second is auto mode - data from buffer are playing forever and interrupt is fired after play buffer. You should use interrupt for re-filling buffer.
Because you only have a single DMA buffer to transfer data, it is recommended that you issue the Play PCM command with a sample count of half of the data buffer. This will fire an interrupt when the buffer is half played, allowing you to refill the segment that was just played, while the DSP continues playing the other half, and so on.
 
==Operations==
==Reading and writing==
Until you write to the DSP write port, check the DSP write port is 0x00. Until reading the DSP read port, check the DSP status for 0x80.
 
==Detecting=Reseting DSP (detecting DSP)===
# Send 1 to DSP reset port
Detecting DSP is really easy. You must send to DSP reset port 1, wait 3 microseconds, send to DSP reset 0 and read DSP read port. If in it is 0xAA, DSP exist.
# Wait 3 microseconds
# Send 0 to DSP reset port
# Now should be in DSP read port 0xAA.
Some SB-compatible cards returned 0xAA on first check but on second check they reported their specific version.
 
===Setting IRQ===
# Send 0x80 to Mixer port
# Send value of your IRQ to Mixer data port
0x01=IRQ 2 0x02=IRQ 5 0x04=IRQ 7 0x08=IRQ 10 Usually is used IRQ 5. You can read this port too for get actual IRQ.
 
===Programming DMA===
You can get more info about programming DMA on page [http://homepages.cae.wisc.edu/~brodskye/sb16doc/sb16doc.html] .
 
Programming 8 bit transfers throught channel 1 (channel number is 1):
# Disable channel by writing to port 0x0A value 0x05 (channel number + 0x04)
# Write value to flip-flop port 0x0C (any value e.g. 1)
# Send transfer mode to 0x0B (0x48 for single mode/0x58 for auto mode + channel number)
# Send page number to 0x83(page port of channel 1) For example if you have sound data at 0x100450, page is 0x10.
# Send low bits of position to port 0x02(addr. port of channel 1) For example(see above) is 0x50.
# Send high bits of position to port 0x02(addr. port of channel 1) For example(see above) is 0x04.
# Send low bits of length of data to port 0x03(count port of channel 1) For example if is length 0x0FFF, send 0xFF
# Send high bits of length of data to port 0x03(count port of channel 1) For example if is length 0x0FFF, send 0x0F
# Enable channel by writing channel number to port 0x0A
 
Programming 16 bit transfers throught channel 5 (channel number is 1 too):
# Disable channel by writing to port 0xD4 value 0x05 (channel number + 0x04)
# Write value to flip-flop port 0xD8 (any value e.g. 1)
# Send transfer mode to 0xD6 (0x48 for single mode/0x58 for auto mode + channel number)
# Send page number to 0x8B(page port of channel 5) For example if you have sound data at 0x100450, page is 0x10.
# Send low bits of position to port 0xC4(addr. port of channel 5) For example(see above) is 0x50.
# Send high bits of position to port 0xC4(pos. port of channel 5) For example(see above) is 0x04.
# Send low bits of length of data to port 0xC6(count port of channel 5) For example if is length 0x0FFF, send 0xFF
# Send high bits of length of data to port 0xC6(count port of channel 5) For example if is length 0x0FFF, send 0x0F
# Enable channel by writing channel number to port 0xD4
 
===Writing transfer mode to DSP===
 
Usually values are 0xB0 for 16 bit playing sound or 0xC0 for 8 bit playing sound.
 
{| {{wikitable}}
! Bit 7-4
! Bit 3
! Bit 2
! Bit 1
! Bit 0
|-
| 0xB=16 bit transfer 0xC=8 bit transfer || 0=playing sound 1=recording sound || 0 || 0=FIFO off 1=FIFO on || 0
|-
|}
 
===Writing type of sound data to DSP===
 
You must write type of sound data after write transfer mode.
 
{| {{wikitable}}
! Bit 7
! Bit 6
! Bit 5
! Bit 4
! Bit 3
! Bit 2
! Bit 1
! Bit 0
|-
| 0 || 0 || 0=mono 1=stereo || 0=unsigned 1=signed || 0 || 0 || 0 || 0
|-
|}
 
==Playing sound==
1.Detect# Reset DSP
2.# Load sound data to memory
# Set master volume
3.Program ISA DMA
# Turn speaker on
4.Set output sample rate
# Program ISA DMA to transfer
5.Write DMA transfer type to DSP
# Set time constant. Notice that the Sound Blaster 16 is able to use sample rates instead of time constants using command 0x41 instead of 0x40. <br> You can calculate the time constant like this: Time constant = 65536 - (256000000 / (channels * sampling rate))
6.Write data lenght to DSP(low byte/high byte)
# Set output sample rate
# Write transfer mode to DSP
# Write type of sound data
# Write data length to DSP(Low byte/High byte) (You must calculate LENGTH-1 e.g. if is your real length 0x0FFF, you must send 0xFE and 0x0F)
 
==Code==
<sourcesyntaxhighlight lang="Casm">
%macro OUTB 2
#define DSP_RESET 0x226
mov dx, %1
#define DSP_READ 0x22A
mov al, %2
#define DSP_WRITE 0x22C
out dx, al
#define DSP_BUFFER 0x22A
%endmacro
#define DSP_STATUS 0x22E
 
#define DSP_INTERRUPT 0x22F
%macro INB 1
mov dx, %1
in al, dx
%endmacro
 
 
;SOUND BLASTER 16 driver in real mode
 
;reset sound blaster
OUTB 0x226, 1 ;reset port
mov ah, 86h
mov cx, 0x0000
mov dx, 0xFFFF
int 15h ;wait
OUTB 0x226, 0 ;reset port
 
;turn speaker on
OUTB 0x22C, 0xD1
 
;DMA channel 1
OUTB 0x0A, 5 ;disable channel 1 (number of channel + 0x04)
OUTB 0x0C, 1 ;flip flop
OUTB 0x0B, 0x49 ;transfer mode
OUTB 0x83, 0x01 ;PAGE TRANSFER (EXAMPLE POSITION IN MEMORY 0x[01]0F04) - SET THIS VALUE FOR YOU
OUTB 0x02, 0x04 ;POSITION LOW BIT (EXAMPLE POSITION IN MEMORY 0x010F[04]) - SET THIS VALUE FOR YOU
OUTB 0x02, 0x0F ;POSITON HIGH BIT (EXAMPLE POSITION IN MEMORY 0x01[0F]04) - SET THIS VALUE FOR YOU
OUTB 0x03, 0xFF ;COUNT LOW BIT (EXAMPLE 0x0FFF) - SET THIS VALUE FOR YOU
OUTB 0x03, 0x0F ;COUNT HIGH BIT (EXAMPLE 0x0FFF) - SET THIS VALUE FOR YOU
OUTB 0x0A, 1 ;enable channel 1
 
;program sound blaster 16
OUTB 0x22C, 0x40 ;set time constant
OUTB 0x22C, 165 ;10989 Hz
OUTB 0x22C, 0xC0 ;8 bit sound
OUTB 0x22C, 0x00 ;mono and unsigned sound data
OUTB 0x22C, 0xFE ;COUNT LOW BIT - COUNT LENGTH-1 (EXAMPLE 0x0FFF SO 0x0FFE) - SET THIS VALUE FOR YOU
OUTB 0x22C, 0x0F ;COUNT HIGH BIT - COUNT LENGTH-1 (EXAMPLE 0x0FFF SO 0x0FFE) - SET THIS VALUE FOR YOU
 
;now transfer start - don't forget to handle irq
void reset_DSP(void) {
</syntaxhighlight>
outb(DSP_RESET, 1);
outb(DSP_RESET, 0);
if(read_DSP()==0xAA) {
sound_blaster=TRUE;
}
}
 
==QEMU support==
void sb16_init(void) {
QEMU is one of the few hypervisors/emulators that support this sound card.
reset_DSP();
To run QEMU with Sound Blaster 16 emulation, use the <code>-soundhw sb16</code> option.
 
'''Warning''': recent versions of QEMU (>= 4.0) have a broken support for
//if DSP doesnt exist
this sound card. See bug: [https://bugs.launchpad.net/qemu/+bug/1873769].
if(sound_blaster==FALSE) {
Briefly, when QEMU's GTK UI is used and audio is playing, you'll experience
return;
the QEMU window freezing. In addition, there will be flickering in the audio
}
as well.
 
===Workarounds===
//get DSP version
write_DSP(0xE1);
sb16_version_major=read_DSP();
sb16_version_minor=read_DSP();
 
# Use an older version of QEMU. With QEMU 2.11 the problem simply does not exist.
}
# Use [https://virt-manager.org virt-manager] which connects to QEMU using Spice. The problem does not exist in this case because QEMU's GTK UI is not used. BUT, configuring QEMU to emulate Sound Blaster 16 through virt-manager requires some tricky settings:
</source>
## Add any sound card to the VM
## In virt-manager's Edit -> Preferences menu check the "Enable XML editing" box.
## Open VM's hardware and after selecting the sound card, click on the XML tab and replace its contents with:<syntaxhighlight lang="xml"><sound model="sb16"/></syntaxhighlight>
# Run the OS using QEMU's <code>-curses</code> option or use QEMU's <code>-kernel</code> and <code>-nographic</code> (serial console): in this case, there will be no "freeze", but there still be some flickering in the audio.
 
==See Also==
===External Links===
* http://www.qzx.com/pc-gpe/
* http://www.inversereality.org/tutorials/sound%20programming/examples/soundblaster16example2.html
* http://homepages.cae.wisc.edu/~brodskye/sb16doc/sb16doc.html
* [http://ftp.lanet.lv/ftp/mirror/x2ftp/msdos/programming/mxinfo/ Some documents at the bottom (.zip)]
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