Playing .wav files on DOSBox's Sound Blaster device

问题

I want to make a program in assembly/8086/masm/dosbox that turns the keyboard into various musical instruments so i need to be able to play some .wav files to produce the required sounds.I am aware of beep char and producing sound via sending frequencies to pc speaker (ports 41h,42h and 61h) but both ways are clearly not going to get me there.

I searched around and found that I need to use int 21h for opening files, knowledge of .wav format and knowledge of sound programming using Sound Blaster.

Unfortunately I couldn't find any helpful documentation on how to use the Sound Blaster in Dosbox (or in general) so kindly if you can help me with my problem of how to play .wav files on dosbox or if you have any workarounds I am all ears ( more accurate eyes).

回答1:

This present a demo program that plays a specific WAV file (to avoid introducing a RIFF parser to the already too-long-for-SO code.
The program has been tested in DOSBox, but a lot of things can go wrong on different configurations.

Finally, I was forced to split the code into two answers.
This is part 1.

Though the question may classify as off-topic¹ I believe it could be a precious resource to have on this site.
So I'm attempting to respond it.

A few notes on the environment:

I've used TASM as the assembler, there is no particular reason behind this choice but childhood memories.
The code should be compatible with MASM.
I'm using DOSBox to emulate a DOS environment.
DOSBox ships with a preconfigured SoundBlaster 16 card.
TASM can be run under DOSBox without any problem.

A scanned version of the TASM 5 manual² is available online.
Though no uncommon syntax has been used, being unfamiliar with the assembler directives makes any code harder to read and understand.
The TASM 5 pack is available online.

Assembling, general source format and debugging

As a matter of convenience, the code developed for this answer can be found on GitHub.
The binary format is the MZ executable with memory model SMALL, one data segment named _DATI³ and one code segment named _CODE.
Each segment is defined multiple times for convenience⁴, both segments are PUBLIC so all these different definitions are merged together by the linker, resulting in just two segments⁵.

The sources target the 8086 as per OP request.

The sources use conditional macro and symbolic values⁶ in order to be configurable, only three values need to be adjusted eventually.
The default values match the default configuration of DOSBox.
We will see the configuration soon.

Due to the not elementary nature of this task, debugging is essential.
To facilitate it, TASM and TLINK can be instructed to generate, and include, debugging symbols.
Coupled with the use of TD debugging is greatly simplified.

Assemble the sources with

tasm /zi sb16.asm
tlink /v sb16.obj

to generates full debugging symbols.
Use td sb16 to debug the program.
Some notes on debugging:

Sometimes DOSBox crashes.
During debugging the DOS environment can be corrupted if the program acts incorrectly or is terminated earlier. Be ready to restart DOSBox often.
Place an int 03h (opcode CC) instruction where you want TD to break. This is handy to debug the ISR.

Soundcard configuration

The SoundBlaster 16 (SB16) had a simple DSP that when filled with digital samples converted them into an analogue output.
To read the samples the card took advantage of a special transfer mode called Direct Memory Access (DMA), the chip that handled such transfers was capable of handling 4x2 in flight data movements.
The SB16 had a jumper, or a switch, to configure the channel to use to read the samplings.

When a block of sampling was over the card requested the attention of the CPU through an interrupt, the chip handling the interrupts had 8x2 request lines.
The SB16 had another jumper to select the Interrupt ReQuest line (IRQ) to use.

Finally, as every legacy device, the SB16 was mapped in the IO address space where it occupied sixteen continuous bytes.
The starting address, a.k.a. base address, of this block was configurable too. A part was fixed and a part was variable, the base address had a form of 2x0h where x was configurable.

All these options are reflected in the DOSBox configuration file.
The program given has been tested with these options⁷:

[sblaster]
sbtype=sb16
sbbase=220
irq=7
dma=1
hdma=5
sbmixer=true
oplmode=auto
oplemu=default
oplrate=44100

Sources configuration

Though this is a premature introduction to the sources, it is handy to present the configuration constants now that we have just seen the DOSBox configurations.

In the file cfg.asm there are these constants

;IO Base
SB16_BASE   EQU 220h

;16-bit DMA channel (must be between 5-7)
SB16_HDMA   EQU 5

;IRQ Number
SB16_IRQ    EQU 7

The values here must reflect the ones present in the DOSBox config file.
Every other constant defined in the file is for the use of the program and not intended to be modified unless you know what you are doing⁸.

The cfg.asm has nothing else of interest and won't be discussed again.

How to play samples

After a long introduction, we are now ready to see how to play a buffer of samples.
A very good and synthetic reference is [available here]tutorial/documentation on the SB16 here.
This answer is basically an implementation of what is written there, with some verbose explanation.

These are the step we will follow:

To playback a buffer of samples the step requested are:

Allocate a buffer that does not cross a 64k physical page boundary

Install an interrupt service routine

Program the DMA controller for background transfer

Set the sampling rate

Write the I/O command to the DSP

Write the I/O transfer mode to the DSP

Write the block size to the DSP (Low byte/high byte)

The goal is to play this WAV file of a Super Mario Bros coin.

Sources organization

There are seven files:

sb16.asm is the main file that includes the others.
It performs the steps above.
cfg.asm contains the configuration constants.
buffer.asm contains the routines for allocating the samples buffer.
data.asm contains the routines that fill the buffer.
This is the file to edit to adapt the source to other goals.
isr.asm contains the routines that set the ISR and the ISR itself.
dma.asm contains the routines that program the DMA.
dsp.asm contains the routines that program the DSP.

In general, the files are short.

The sample buffer

The high-level process is as follow: the card is given a buffer to read, when done it triggers an interrupt and stops; the software then update the buffer and restart the playback.
The drawback with this method is that it introduces pauses in the playback that present themselves as audible "clicks".
The DMA and the DSP support a mode called auto-initialize where, when the end of the buffer is reached, the transfer and the playback start over from the start.
This is good for a cyclic static buffer but won't help for an ever-updating buffer.

The trick is to program the DMA to transfer a block twice as large as the block the DSP is programmed to read. This will make the card generate an interrupt at the middle of the buffer.
The software will then resume the playback immediately and then update the half just read. This is explained in the diagram below.

How big should the buffer be?

I have chose a size of 1/100 sec at 44100 samples per second, mono, 16-bit per sample. This is 441 samples times 1 audio channel times 2 bytes per sample.
This is the block size. Since we have two blocks, the buffer size should be twice as much.
In practice, it is four times as much (in the end, it is about 3.5 KiB).

The big problem with the buffer is that it must not cross a physical 64KiB boundary⁹.
Note that this is not the same as not crossing a 64KiB logical boundary (which is impossible without changing segment).

I couldn't find a suitable allocation routine in the Ralf Brown Interrupt List, so I proceeded by abstracting the behaviour in two routines.

AllocateBuffer that must set the variables bufferOffset and bufferSegment with the far pointer to the allocated buffer of size at least BLOCK_SIZE * 2.
Upon return, if CF it means the procedure failed.
FreeBufferIfAllocated that is called to free the buffer. It is up to this procedure to check if a buffer was effectively allocated or not.

The default implementation statically allocates in the data segment a buffer that is twice as needed, as said.
My reasoning was that if this unmoveable buffer crosses a 64KiB boundary than it is split into two halves, L and H, and it is true that L + H = BLOCK_SIZE * 2 * 2.
Since the worst case scenario is when L = H, i.e. the buffer is split in the middle, the double size gives a size of BLOCK_SIZE * 2 * 2 / 2 = BLOCK_SIZE * 2 in the worst case scenario for both L and H.
This guarantees us that we can always find a half as large as BLOCK_SIZE * 2, which is what we needed.

The AllocateBuffer just find an appropriate half and set the value of the far pointer mentioned above. FreeBufferIfAllocated does nothing.

Note that by "buffer" I mean two "blocks" and a "block" is the unit of playback.

What format should the buffer use?

To keep the things simple, the DSP is programmed to playback 16-bit mono samplings.
However, the procedures that fill the blocks have been abstracted into data.asm.

UpdateBuffer is called by the ISR to update a block.
The parameters are

AX = Block number (Either 0 or 1) BX = Block mask (0 for block 0, 0ffffh for block 1)

They are used to compute the offset into the buffer with this code
```
;Set ES:DI to point to start of the current block

mov di, WORD PTR [bufferSegment]
mov es, di
mov di, BLOCK_SIZE
and di, bx
add di, WORD PTR [bufferOffset]
```
The rest of the procedure read a block of samples from the WAV file.
If the file has ended, the file pointer is reset back to the beginning to implement a cycling playback.

BEWARE You are called in an ISR context, while the ACK and the EOI have already been issued, you must not clobber any register.
Failing to respect this rule will result in difficult to understand bugs and possibly freezes.
- InitBuffer is called one at the beginning to initialize the buffer if needed.
  The current implementation opens the file coin.wav¹⁰, read the sample rate and set the file pointer to the data section.
  This procedure uses the CF to signal an error. If the CF is set, an error has been encountered and DX holds a pointer to a $ terminated string that will be printed.
- FinitBuffer used at the end to free the buffer resources.
  The buffer memory itself is freed as said above.
  This is called even if InitBuffer fails.

We will talk about the WAV reading below.

Installing the ISR

I assume you are familiar with the IVT.
I suggest reading about the twos 8259A PIC used to routes IRQs.

In shorts:

There are 15 IRQ lines, from 0 to 15, 2 excluded.
An IRQ line must be enabled (unmasked) before the use.
After an IRQ has been served, an End of Interrupt (EOI) must be sent to the PIC that served it. IRQs above 7 are served by both PICs.
IRQ 0-7 are mapped to interrupt numbers 08h-0fh, IRQ 8-15 to 70h-78h

The file isr.asm is very short.
The routine SwapISRs swap the current ISR pointer for the IRQ of the SB16 with a local pointer.
Initially, this pointer points to the ISR Sb16Isr, so that the first call to SwapISRs will install our ISR.
The second call will restore the original one.

Sb16Isr does a few things:

It acknowledges the IRQ to the SB16 (more on this later).
It files the EOI to the PIC(s).
It calls UpdateBuffer.
It updates the block number and block mask passed to the routine above.

NOTE SwapISRs also toggles the bit for the IRQ mask. It assumes that the IRQ is masked at the beginning of the program. You may want to change this to a more robust setting (or restart DOSBox if you abruptly interrupt the program).

Programming the DMA controller

The SB16 was an ISA card, it couldn't read the memory directly.
To solve this problem the DMA chip, 8357 was invented.
It had four channels, independently configurable, that when triggered performed a read from the memory to the ISA bus or vice-versa.
There were two DMA controllers, the first one handled only 8-bit data transfers and channels 0-3.
The second one used 16-bit data transfers and handled the channels 4-7.

We are going to use the 16-bit transfers so, the DMA channel must be one of 5-7 (channel four is a bit special).
The SB16 can also use 8-bit transfers, so it has two configurations for the DMA channel: one for the 8-bit moves and one for the 16-bit moves.

Each channel, but channel four, has three parts:

A 16-bit start address.
A 16-bit counter for the size.
An 8-bit page number.

The address is a physical address (linear)! So in theory only the first 64KiB were accessible.
The page number was used as the upper part of the address.
However, the counter logic is still 16-bit, so the pointer to the data to read/write still wrap around at 64 KiB boundaries (should be 128 KiB for 16-bit).

The dma.asm files contain a single routine SetDMA that given the logical start address and the size, program the DMA.
There isn't anything esoteric here besides a few arithmetic to compute the value to use.

The mode is Single mode and auto-initialization is on.
The document about the SB16 programming liked at the beginning has a very clear step-by-step procedure on this.

Programming the DSP

The SB16 IO layout was as follow:

ADDR    READ                        WRITE   

2x6h    DSP Reset*                  DSP reset**

2xAh    DSP Read***  

2xCh    DSP Write                   DSP write (command and data)
        (bit7 set if ok to write)

2xEh    DSP Read Status****
        (bit7 set if ok to read)

2xFh    DSP 16-bit interrupt acknowledge 

* bit 7 set after the reset completes
** toggle bit 7, with a 3us interval between setting and clearing, to start a reset
*** Wait for reading a 0AAh after a reset
**** Also used to ACK 8-bit IRQs

The file dsp.asm contains the basic routines ResetDSP, WriteDSP and ReadDSP that performs a reset, write a byte to the DSP after waiting for right conditions, read a byte from the DSP.

The DSP is used through commands.

To set the sampling of the playback use the command 41h, followed by the low byte of the sampling frequency and then by the high byte.
The routine SetSampling takes the sampling frequency in AX and set it.
To playback use the command b6h, followed by a mode byte and then by the block length (two bytes, low byte first).
The routine StartPlayback takes the sampling frequency in AX, the mode byte in BL and the size in CX and start a playback (after setting the sample rate).
Note that the DSP doesn't need to know the address of the buffer, it just triggers the channel request pin of the DMA and it will have the data on the bus.
It is the DMA that have to know where the buffer is.
To stop a playback use the command d5h.
StopPlayback does this.

Playing the WAV file

What the demo program do is playing the coin.wav file.
This is file is specific it is a 16-bit mono file.

The demo program doesn't parse the full RIFF format (you can see this nice page, it is hardwired to work with that specific file.
Though any file with identical format, yet different data, should do.

After the steps introduce at the beginning, the program simply wait for a keystroke.
After that it performs all the de-initializations (including stopping the playback) and exit.

To continue from here, you have "only" to properly implement the routine in data.asm.
It should be straightforward to make each key plays a different file.

If the number of file is small I would open all the files in InitBuffer, then in sb16.asm implement a loop liked

xor ah, ah
int 16h

cmp al, ...
je ...

cmp al, ...
je ...

where each jump gets the file handle to play. (a lookup table would be better).
Then:

Reset the file pointer of the file to play to the start of the samples.
xchg the new file pointer with fileHandle (used by UpdateBuffer).

I leave to you how to make the playback stop when the key is released and resume when it is pressed.

Code

sb16.asm

.386
.387
.MODEL SMALL

.STACK


INCLUDE cfg.asm
INCLUDE buffer.asm


_DATI SEGMENT PARA PUBLIC 'DATA' USE16

 ;This is the segment to the buffer for the sampling

 samplingBuffer   dw 0 

 ;Strings

 strErrorBuffer   db "Cannot allocate or find a buffer for the samplings :(", 24h

 strPressAnyKey   db "Press any key to exit", 13, 10, 24h
 strBye           db "Sound should stop now", 13, 10, 24h

_DATI ENDS

INCLUDE data.asm
INCLUDE isr.asm
INCLUDE dsp.asm
INCLUDE dma.asm

_CODE SEGMENT PARA PUBLIC 'CODE' USE16
 ASSUME CS:_CODE, DS:_DATI, ES:_DATI

__START__:

 ;Basic initialization

 mov ax, _DATI
 mov ds, ax



 ;S E T   T H E   N E W   I S R

 call SwapISRs



 ;A L L O C A T E   T H E   B U F F E R

 call AllocateBuffer
 mov dx, OFFSET strErrorBuffer
jc _error



 ;I N I T   T H E   B U F F E R

 call InitBuffer
jc _finit_buffer


 ;S E T U P   D M A

 mov si, WORD PTR [bufferSegment]
 mov es, si
 mov si, WORD PTR [bufferOffset]
 mov di, BLOCK_SIZE * 2
 call SetDMA



 ;S T A R T   P L A Y B A C K

 call ResetDSP

 mov ax, WORD PTR [sampleRate]            ;Sampling
 mov bx, FORMAT_MONO OR FORMAT_SIGNED     ;Format
 mov cx, BLOCK_SIZE                       ;Size
 call StartPlayback



 ;W A I T

 mov ah, 09h
 mov dx, OFFSET strPressAnyKey
 int 21h

 xor ah, ah
 int 16h



 ;S T O P

 call StopPlayback

 mov dx, OFFSET strBye

_finit_buffer:

 ;F R E E   B L O C K   R E S O U R C E S
 call FinitBuffer


 ;E R R O R   H A N D L I N G

 ;When called DX points to a string

_error:
 ;R E S T O R E   T H E   O L D   I S R s

 call SwapISRs



 call FreeBufferIfAllocated

 mov ah, 09h
 int 21h

 ;E N D

_end:
 mov ax, 4c00h
 int 21h

_CODE ENDS

END __START__

cfg.asm

 ;These are the only configurable constants

 ;IO Base
 SB16_BASE   EQU 220h

 ;16-bit DMA channel (must be between 5-7)
 SB16_HDMA   EQU 5

 ;IRQ Number
 SB16_IRQ    EQU 7

 ;These a computed values, don't touch them if you don't know what
 ;you are doing

 ;REGISTER NAMES

 REG_DSP_RESET      EQU SB16_BASE + 6
 REG_DSP_READ       EQU SB16_BASE + 0ah
 REG_DSP_WRITE_BS   EQU SB16_BASE + 0ch
 REG_DSP_WRITE_CMD  EQU SB16_BASE + 0ch
 REG_DSP_WRITE_DATA EQU SB16_BASE + 0ch
 REG_DSP_READ_BS    EQU SB16_BASE + 0eh
 REG_DSP_ACK        EQU SB16_BASE + 0eh
 REG_DSP_ACK_16     EQU SB16_BASE + 0fh

 ;DSP COMMANDS

 DSP_SET_SAMPLING_OUTPUT   EQU 41h
 DSP_DMA_16_OUTPUT_AUTO    EQU 0b6h
 DSP_STOP_DMA_16           EQU 0d5h

 ;DMA REGISTERS

 REG_DMA_ADDRESS    EQU 0c0h + (SB16_HDMA - 4) * 4
 REG_DMA_COUNT      EQU REG_DMA_ADDRESS + 02h

 REG_DMA_MASK       EQU 0d4h
 REG_DMA_MODE       EQU 0d6h
 REG_DMA_CLEAR_FF   EQU 0d8h


 IF SB16_HDMA - 5
    REG_DMA_PAGE       EQU 8bh      
 ELSE
    IF SB16_HDMA - 6
       REG_DMA_PAGE       EQU 89h
    ELSE
       REG_DMA_PAGE       EQU 8ah
    ENDIF
 ENDIF

 ;ISR vector
 ISR_VECTOR            EQU ((SB16_IRQ SHR 3) * (70h - 08h) + (SB16_IRQ AND 7) + 08h) * 4

 PIC_DATA       EQU (SB16_IRQ AND 8) + 21h
 PIC_MASK               EQU 1 SHL (SB16_IRQ AND 7)

dma.asm

.8086
.MODEL SMALL


_CODE SEGMENT PARA PUBLIC 'CODE' USE16
 ASSUME CS:_CODE


 ;ES = buffer segment
 ;SI = buffer offset
 ;DI = count
 SetDMA:
  push dx
  push ax
  push cx
  push bx
  push si

  ;Disable SB16 DMA channel

  mov dx, REG_DMA_MASK
  mov al, 4 + SB16_HDMA MOD 4
  out dx, al

  ;Clear counter FF

  mov dx, REG_DMA_CLEAR_FF
  out dx, al

  ;Set transfert mode

  mov dx, REG_DMA_MODE
  mov al, 58h + SB16_HDMA MOD 4
  out dx, al


  ;Set address (in WORDs)

  ;SSSS SSSS SSSS SSSS 0000
  ;0SSS SSSS SSSS SSSS S000

  mov bx, es
  shr bx, 0dh        ;BL = addr[20:16]  
  mov cx, es
  shl cx, 3          ;CX = addr[15:0]
  shr si, 1
  add cx, si
  adc bx, 0             

  mov dx, REG_DMA_ADDRESS
  mov al, cl
  out dx, al
  mov al, ch
  out dx, al

  mov dx, REG_DMA_PAGE
  mov al, bl
  out dx, al

  ;Set count

  mov ax, di
  shr ax, 1

  mov dx, REG_DMA_COUNT
  out dx, al
  mov al, ah
  out dx, al

  ;Enable DMA channel

  mov dx, REG_DMA_MASK
  mov al, SB16_HDMA MOD 4
  out dx, al

  pop si
  pop bx
  pop cx
  pop ax
  pop dx

  ret

_CODE ENDS

isr.asm

.8086
.MODEL SMALL

_DATI SEGMENT PARA PUBLIC 'DATA' USE16

 ;This is a pointer to the ISR we will install 

 nextISR         dw OFFSET Sb16Isr
                 dw _CODE

 ;This is the internal status managed by the ISR

 blockNumber     dw 0
 blockMask       dw 0

_DATI ENDS

_CODE SEGMENT PARA PUBLIC 'CODE' USE16
 ASSUME CS:_CODE, DS:_DATI, ES:_DATI

 ;Swaps two far pointers

 ;DS:SI = ptr to ptr1
 ;ES:DI = ptr to ptr2
 SwapFarPointers:
  push bx

  mov bx, WORD PTR [si]
  xchg WORD PTR es:[di], bx
  mov WORD PTR [si], bx

  mov bx, WORD PTR [si+02h]
  xchg WORD PTR es:[di+02h], bx
  mov WORD PTR [si+02h], bx

  pop bx
  ret  

 ;Swaps the ISR vector of the IRQ of the card with a saved value

 SwapISRs:
  push es
  push si
  push di
  push dx
  push ax

  cli

  mov si, OFFSET nextISR
  xor di, di
  mov es, di
  mov di, ISR_VECTOR
  call SwapFarPointers

  sti

  ;Toggle PIC mask bit
  mov dx, PIC_DATA
  in al, dx
  xor al, PIC_MASK
  out dx, al

  pop ax
  pop dx
  pop di
  pop si
  pop es
  ret  


 ;This is the ISR

 Sb16Isr:
  push ax
  push dx
  push ds
  push es

  ;Ack IRQ to SB16

  mov dx, REG_DSP_ACK_16
  in al, dx

  ;EOI to PICs

  mov al, 20h
  out 20h, al

IF SB16_IRQ SHR 3 
  out 0a0h, al
ENDIF

  mov ax, _DATA
  mov ds, ax

  mov ax, WORD PTR [BlockNumber]
  mov bx, WORD PTR [BlockMask]  
  call UpdateBuffer

  not bx
  inc ax
  and al, 01h

  mov WORD PTR [BlockNumber], ax
  mov WORD PTR [BlockMask], bx

  pop es
  pop ds
  pop dx
  pop ax
  iret


_CODE ENDS

data.asm

.8086
.MODEL SMALL

_DATI SEGMENT PARA PUBLIC 'DATA' USE16

  strWaveFile           db "coin.wav", 0
  strFileNotFound       db "File not found!", 24h
  strFileError          db "Error while reading WAV file!", 24h

  fileHandle            dw 0

  sampleRate            dw 0

_DATI ENDS

_CODE SEGMENT PARA PUBLIC 'CODE' USE16
 ASSUME CS:_CODE, DS:_DATI, ES:_DATI

 ;This is called to update the block given


 ;AX = Block number (Either 0 or 1)
 ;BX = Block mask (0 for block 0, 0ffffh for block 1)
 UpdateBuffer:
  push es
  push di
  push bx
  push ax
  push si
  push cx
  push dx

  ;Set ES:DI to point to start of the current block

  mov di, WORD PTR [bufferSegment]
  mov es, di
  mov di, BLOCK_SIZE
  and di, bx
  add di, WORD PTR [bufferOffset]

  ;Read from file

  push ds

  mov ax, es
  mov ds, ax
  mov dx, di

  mov ah, 3fh
  mov bx, WORD PTR [fileHandle]
  mov cx, BLOCK_SIZE
  int 21h

  pop ds  

  ;Check if EOF

  cmp ax, BLOCK_SIZE
  je _ub_end

  mov ax, 4200h
  mov bx, WORD PTR [fileHandle]
  xor cx, cx
  mov dx, 44d
  int 21h

 _ub_end:
  pop dx
  pop cx
  pop si
  pop ax
  pop bx
  pop di
  pop es
  ret

 ;This is called to initialize both blocks
 ;Set CF on return (and set DX to the offset of a string) to show an error and exit
 InitBuffer:
  push ax
  push bx

  ;finit

  ;xor ax, ax
  ;mov bx, ax
  ;call UpdateBuffer

  ;inc al
  ;not bx
  ;call UpdateBuffer


  mov ax, 3d00h
  mov dx, OFFSET strWaveFile
  int 21h

  mov dx, OFFSET strFileNotFound
  mov WORD PTR [fileHandle], ax
 jc _ib_end

  ;Read sample rate

  mov bx, ax
  mov ax, 4200h
  xor cx, cx
  mov dx, 18h
  int 21h

  mov dx, OFFSET strFileError
 jc _ib_end

  mov ah, 3fh
  mov bx, WORD PTR [fileHandle]
  mov cx, 2
  mov dx, OFFSET sampleRate
  int 21h

  mov dx, WORD PTR [sampleRate]             ;DEBUG

  mov dx, OFFSET strFileError
 jc _ib_end

  ;Set file pointer to start of data

  mov ax, 4200h
  mov bx, WORD PTR [fileHandle]
  xor cx, cx
  mov dx, 44d
  int 21h


 _ib_end:
  pop bx
  pop ax
  ret


 ;Closed to finalize the buffer before exits

 FinitBuffer:
  push ax
  push bx
  push dx

  mov bx, WORD PTR [fileHandle]
  test bx, bx
 jz _fib_end

  mov ah, 3eh
  int 21h

 _fib_end:
  pop dx
  pop bx
  pop ax
  ret

_CODE ENDS

¹ For example because it asks for a non-trivial amount of code or for a resource.

² Beware that the Table Of Content has some pages switched.

³_DATA is already defined.

⁴ Each source file redefine those segments if used.

⁵ The symbols _DATI and _CODE can be used to denote the segment part of the starting address of the final segments.

⁶ I don't remember the exact technical name for the EQU values.

⁷ These values are DOSBox defaults but be sure to check the config file anyway.

⁸ Specially because TASM lacks supports for a lot of conditional and a bit of bit-arithmetic is needed to set some value.

⁹ This should be 128KiB for 16-bit DMA, which we are using, but I don't remember exactly and didn't want to experiment.

¹⁰ Beware of DOS limitations on file names.

回答2:

This present a demo program that plays a specific WAV file (to avoid introducing a RIFF parser to the already too-long-for-SO code. The program has been tested in DOSBox, but a lot of things can go wrong on different configurations.

Finally, I was forced to split the code into two answers.
This is part 2.

dsp.asm

.8086
.MODEL SMALL

 FORMAT_MONO     EQU 00h
 FORMAT_STEREO   EQU 20h
 FORMAT_SIGNED   EQU 10h
 FORMAT_UNSIGNED EQU 00h


_CODE SEGMENT PARA PUBLIC 'CODE' USE16
 ASSUME CS:_CODE


 ResetDSP:
  push ax
  push dx

  ;Set reset bit

  mov dx, REG_DSP_RESET
  mov al, 01h
  out dx, al

  ;Wait 3 us

  in al, 80h
  in al, 80h
  in al, 80h

  ;Clear reset bit

  xor al, al
  out dx, al

  ;Poll BS until bit 7 is set

  mov dx, REG_DSP_READ_BS

 _rd_poll_bs:
  in al, dx
  test al, 80h
 jz _rd_poll_bs

  ;Poll data until 0aah

  mov dx, REG_DSP_READ

 _rd_poll_data:
  in al, dx
  cmp al, 0aah
 jne _rd_poll_data

  pop dx
  pop ax
  ret



 ;AL = command/data
 WriteDSP:
  push dx
  push ax

  mov dx, REG_DSP_WRITE_BS

 _wd_poll:
  in al, dx
  test al, 80h
 jz _wd_poll

  pop ax

  mov dx, REG_DSP_WRITE_DATA
  out dx, al

  pop dx
  ret


 ;Return AL 
 ReadDSP:
  push dx

  mov dx, REG_DSP_READ_BS

 _rdd_poll:
  in al, dx
  test al, 80h
 jz _rdd_poll

  pop ax

  mov dx, REG_DSP_READ
  in al, dx

  pop dx
  ret


 ;AX = sampling
 SetSampling:
  push dx

  xchg al, ah

  push ax

  mov al, DSP_SET_SAMPLING_OUTPUT
  call WriteDSP

  pop ax

  call WriteDSP

  mov al, ah
  call WriteDSP

  pop dx
  ret


 ;Starts a playback

 ;AX = Sampling
 ;BL = Mode
 ;CX = Size
 StartPlayback:

  ;Set sampling

  call SetSampling

  ;Start playback command

  mov al, DSP_DMA_16_OUTPUT_AUTO
  call WriteDSP
  mov al, bl            
  call WriteDSP                            ;Format 
  mov al, cl
  call WriteDSP                            ;Size (Low)
  mov al, ch   
  call WriteDSP                            ;Size (High)

  ret

 ;Stops the playback

 StopPlayback:
   push ax

   mov al, DSP_STOP_DMA_16
   call WriteDSP

   pop ax
   ret

_CODE ENDS

buffer.asm

.8086
.MODEL SMALL


 ;Block size is 1/100 of a second at 44100 samplings per seconds

 BLOCK_SIZE      EQU 44100 / 100 * 2

 ;Buffer size allocated, it is twice the BLOCK_SIZE because there are two blocks.
 ;Size is doubled again so that we are sure to find an area that doesn't cross a
 ;64KiB boundary
 ;Total buffer size is about 3.5 KiB

 BUFFER_SIZE     EQU  BLOCK_SIZE * 2 * 2

_DATI SEGMENT PARA PUBLIC 'DATA' USE16

 ;This is the buffer

 buffer            db BUFFER_SIZE DUP(0)


 bufferOffset      dw OFFSET buffer
 bufferSegment     dw _DATI

_DATI ENDS


_CODE SEGMENT PARA PUBLIC 'CODE' USE16
 ASSUME CS:_CODE, DS:_DATI


 ;Allocate a buffer of size BLOCK_SIZE * 2 that doesn't cross
 ;a physical 64KiB
 ;This is achieved by allocating TWICE as much space and than
 ;Aligning the segment on 64KiB if necessary


 AllocateBuffer:
  push bx
  push cx
  push ax
  push dx

  ;Compute linear address of the buffer

  mov bx, _DATI
  shr bx, 0ch
  mov cx, _DATI
  shl cx, 4
  add cx, OFFSET buffer
  adc bx, 0                                 ;BX:CX = Linear address



  ;Does it starts at 64KiB?

  test cx, cx
 jz _ab_End                                ;Yes, we are fine

  mov dx, cx
  mov ax, bx

  ;Find next start of 64KiB

  xor dx, dx
  inc ax

  push ax
  push dx

  ;Check if next boundary is after our buffer

  sub dx, cx
  sub ax, bx

  cmp dx, BUFFER_SIZE / 2

  pop dx
  pop ax

 jae _ab_end



  mov bx, dx
  and bx, 0fh
  mov WORD PTR [bufferOffset], bx

  mov bx, ax
  shl bx, 0ch
  shr dx, 04h
  or bx, dx
  mov WORD PTR [bufferSegment], bx

 _ab_end:
  clc

  pop dx
  pop ax
  pop cx
  pop bx

  ret


 ;Free the buffer

 FreeBufferIfAllocated:

  ;Nothing to do

  ret




_CODE ENDS

来源：https://stackoverflow.com/questions/41359112/playing-wav-files-on-dosboxs-sound-blaster-device

标签

audio

assembly

wav

x86-16

microprocessors