*** ScR *** *** NOTES *** * MAN! The non-table 14 bit routines sound terrible! * There is still room for improvements in the softint's. For example, * try if reducing chip memory accesses increase speed. Not high priority, though. * This driver should probably have a DMA play mode for 1-4 stereo channels. * Sampler routines should be added. * MINPER should not be fixed! Examine if the current scan mode is doublescan. PALFREQ EQU 3546895 NTSCFREQ EQU 3579545 MINFREQ EQU 7000 MAXPER EQU PALFREQ/MINFREQ MINPER EQU 62 ;------------ auto wo AHI:libs/ahi/paula.audio\ incdir include: include hardware/all.i include exec/exec.i include graphics/gfxbase.i include utility/utility.i include utility/hooks.i include devices/audio.i include dos/dos.i include lvo/exec_lib.i include lvo/utility_lib.i include lvo/dos_lib.i include libraries/ahi.i include libraries/ahi_sub.i include macros.i * paula.audio extra tags AHIDB_Paula14Bit EQU AHIDB_UserBase+0 * Boolean AHIDB_PaulaTable EQU AHIDB_UserBase+1 * Boolean MODE EQU A4000!MC040 INTF_AUDIO EQU INTF_AUD0!INTF_AUD1!INTF_AUD2!INTF_AUD3 AUDINT_OFF MACRO move.w #INTF_AUDIO,CUSTOM+INTENA ENDM AUDINT_ON MACRO move.w #INTF_SETCLR!INTF_AUD0,CUSTOM+INTENA ENDM * paulaBase (private) STRUCTURE paulaBase,LIB_SIZE UBYTE pb_Flags UBYTE pb_Pad1 UWORD pb_Pad2 APTR pb_SysLib ULONG pb_SegList APTR pb_GfxLib APTR pb_UtilLib APTR pb_DosLib LABEL paulaBase_SIZEOF * paula (private) ahiac_DriverData points to this structure. STRUCTURE paula,0 UBYTE p_Flags UBYTE p_Pad UWORD p_DisableCount APTR p_audioport APTR p_audioreq ULONG p_audiodev ; UWORD p_OldDMACON ; UWORD p_OldINTENA APTR p_Interrupt APTR p_NoInterrupt ; APTR p_OldInterrupt APTR p_DMAbuffer *** IS_DATA for Audio Interrupt (do not change order!) LABEL p_AudIntData APTR p_SoftInt UWORD p_AudLen ;don't change order UWORD p_AudPer ; ULONG p_DoubleBufferFlag APTR p_AudPtr1A APTR p_AudPtr2A APTR p_AudPtr3A APTR p_AudPtr4A APTR p_AudPtr1B APTR p_AudPtr2B APTR p_AudPtr3B APTR p_AudPtr4B *** IS_DATA for Software Interrupt (do not change order!) LABEL p_SoftIntData APTR p_PlayerHook ULONG p_Reserved APTR p_AudioCtrlP FPTR p_PlayerEntry ;p_PlayerHook->h_Entry ULONG p_LoopTimes APTR p_MixHook APTR p_Mixbuffer APTR p_AudioCtrlM FPTR p_MixEntry ;p_MixHook->h_Entry APTR p_AudPtrs APTR p_CalibrationTable STRUCT p_CalibrationArray,256 LABEL paula_SIZEOF * p_Flags ; BITDEF P,INT,7 ;interrup handler added BITDEF P,14BIT,0 PB_STEREO EQU AHIACB_STEREO ;=2 PF_STEREO EQU AHIACF_STEREO Start: moveq #-1,d0 rts VERSION EQU 1 REVISION EQU 5 DATE MACRO dc.b "14.3.96" ENDM VERS MACRO dc.b "paula 1.5" ENDM VSTRING MACRO VERS dc.b " (" DATE dc.b ")",13,10,0 ENDM VERSTAG MACRO dc.b 0,"$VER: " VERS dc.b " (" DATE dc.b ")",0 ENDM RomTag: DC.W RTC_MATCHWORD DC.L RomTag DC.L EndCode DC.B RTF_AUTOINIT DC.B VERSION ;version DC.B NT_LIBRARY DC.B 0 ;pri DC.L LibName DC.L IDString DC.L InitTable LibName: dc.b "paula.audio",0 IDString: VSTRING gfxName: GRAPHICSNAME utilName: UTILITYNAME dosName: DOSNAME cnop 0,2 InitTable: DC.L paulaBase_SIZEOF DC.L funcTable DC.L dataTable DC.L initRoutine funcTable: dc.l Open dc.l Close dc.l Expunge dc.l Null * dc.l AHIsub_AllocAudio dc.l AHIsub_FreeAudio dc.l AHIsub_Disable dc.l AHIsub_Enable dc.l AHIsub_Start dc.l AHIsub_Update dc.l AHIsub_Stop dc.l AHIsub_SetVol dc.l AHIsub_SetFreq dc.l AHIsub_SetSound dc.l AHIsub_SetEffect dc.l AHIsub_LoadSound dc.l AHIsub_UnLoadSound dc.l AHIsub_GetAttr dc.l -1 dataTable: INITBYTE LN_TYPE,NT_LIBRARY INITLONG LN_NAME,LibName INITBYTE LIB_FLAGS,LIBF_SUMUSED!LIBF_CHANGED INITWORD LIB_VERSION,VERSION INITWORD LIB_REVISION,REVISION INITLONG LIB_IDSTRING,IDString DC.L 0 initRoutine: movem.l d1/a0/a1/a5/a6,-(sp) move.l d0,a5 move.l a6,pb_SysLib(a5) move.l a0,pb_SegList(a5) lea gfxName(pc),a1 moveq #0,d0 call OpenLibrary move.l d0,pb_GfxLib(a5) bne.b .gfxOK ALERT AG_OpenLib!AO_GraphicsLib moveq #0,d0 bra.b .exit .gfxOK lea utilName(pc),a1 moveq #0,d0 call OpenLibrary move.l d0,pb_UtilLib(a5) bne.b .utilOK ALERT AG_OpenLib!AO_UtilityLib moveq #0,d0 bra.b .exit .utilOK lea dosName(pc),a1 moveq #0,d0 call OpenLibrary move.l d0,pb_DosLib(a5) bne.b .dosOK ALERT AG_OpenLib!AO_DOSLib moveq #0,d0 bra.b .exit .dosOK move.l a5,d0 .exit movem.l (sp)+,d1/a0/a1/a5/a6 rts Open: moveq #0,d0 addq.w #1,LIB_OPENCNT(a6) bclr.b #LIBB_DELEXP,pb_Flags(a6) move.l a6,d0 .exit rts Close: moveq #0,d0 subq.w #1,LIB_OPENCNT(a6) bne.b .exit btst.b #LIBB_DELEXP,pb_Flags(a6) beq.b .exit bsr.b Expunge .exit rts Expunge: movem.l d1/d2/a0/a1/a5/a6,-(sp) move.l a6,a5 move.l pb_SysLib(a5),a6 tst.w LIB_OPENCNT(a5) beq.b .notopen bset.b #LIBB_DELEXP,pb_Flags(a5) moveq #0,d0 bra.b .Expunge_end .notopen move.l pb_GfxLib(a5),a1 call CloseLibrary move.l pb_UtilLib(a5),a1 call CloseLibrary move.l pb_DosLib(a5),a1 call CloseLibrary move.l pb_SegList(a5),d2 move.l a5,a1 call Remove moveq #0,d0 move.l a5,a1 move.w LIB_NEGSIZE(a5),d0 sub.l d0,a1 add.w LIB_POSSIZE(a5),d0 call FreeMem move.l d2,d0 .Expunge_end movem.l (sp)+,d1/d2/a0/a1/a5/a6 rts Null: moveq #0,d0 rts * BeginIO(ioRequest)(a1) (From amiga.lib) BeginIO: move.l a1,a0 ;probably not neccesary push a6 move.l IO_DEVICE(a1),a6 jsr -30(a6) pop a6 rts ******* .audio/AHIsub_AllocAudio ************************************ * * NAME * AHIsub_AllocAudio -- Allocates and initializes the audio hardware. * * SYNOPSIS * result = AHIsub_AllocAudio( tags, audioctrl); * D0 A1 A2 * * ULONG AHIsub_AllocAudio( struct TagItem *, struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * Allocate and initialize the audio hardware. Decide if and how you * wish to use the mixing routines provided by 'ahi.library', by looking * in the AHIAudioCtrl structure and parsing the tag list for tags * you support; * * 1) Use mixing routines with timing: * You will need to be able to play any number of samples from * about 80 up to 65535 with low overhead. * · Update AudioCtrl->ahiac_MixFreq to nearest value that your * hardware supports. * · Return AHISF_MIXING|AHISF_TIMING. * 2) Use mixing routines without timing: * If there is much overhead involved with starting playback, * or the hardware can't play samples with any length, use this * alternative and provide timing yourself. * · Update AudioCtrl->ahiac_MixFreq to nearest value that your * hardware supports. * · Store the number of samples to mix each pass in * AudioCtrl->ahiac_BuffSamples. * · Return AHISF_MIXING * 3) Don't use mixing routines: * If your hardware can handle everying without using the CPU to * mix the channels, you tell 'ahi.library' this by returning NULL. * * If you can handle stereo output from the mixing routines, or you * have a DMA/DSP stereo card, also set bit AHISB_KNOWSTEREO. * If you can handle hifi (32 bit) output from the mixing routines, if * you have a 12 bit or better DMA card, or if a DSP card which uses * non-lossy mixing routines (perhaps with interpolation etc), also set * bit AHISB_KNOWHIFI. * * INPUTS * tags - pointer to a taglist. * audioctrl - pointer to an AHIAudioCtrl structure. * * TAGS * The tags are from the audio database (AHIDB_#? in ), * NOT the tag list the user called ahi.library/AHI_AllocAudio() with. * * RESULT * Flags, defined in . * * EXAMPLE * * NOTES * You don't have to clean up on failure, AHIsub_FreeAudio() will * allways be called. * * BUGS * * SEE ALSO * AHIsub_FreeAudio() * ***************************************************************************** * * AHIsub_AllocAudio: pushm d2-d7/a2-a6 move.l a6,a5 move.l a1,d3 move.l pb_SysLib(a5),a6 move.l #paula_SIZEOF,d0 move.l #MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,ahiac_DriverData(a2) beq.w .error_nopaula move.l d0,a3 move.l a2,p_AudioCtrlP(a3) ;player Hook move.l a2,p_AudioCtrlM(a3) ;mixer Hook moveq #0,d2 move.l pb_UtilLib(a5),a6 move.l #AHIDB_Paula14Bit,d0 moveq #0,d1 move.l d3,a0 ;tag list call GetTagData tst.l d0 beq.b .no14bit moveq #PF_14BIT,d2 .no14bit move.l ahiac_Flags(a2),d0 and.b #PF_STEREO,d0 ;same as AHIACF_STEREO or.b d2,d0 move.b p_Flags(a3),d1 and.b #~(PF_STEREO|PF_14BIT),d1 or.b d0,d1 move.b d1,p_Flags(a3) move.l #AHIDB_PaulaTable,d0 moveq #0,d1 move.l d3,a0 ;tag list call GetTagData tst.l d0 beq.w .notable * Load 'ENV:CyberSound/SoundDrivers/14Bit_Calibration', allocate * and initialize the table. * FIXIT: The calibration file should move to a special hunk in * 'DEVS:AudioModes/PAULA'. move.l pb_DosLib(a5),a6 lea .calibname(pc),a0 move.l a0,d1 move.l #MODE_OLDFILE,d2 call Open move.l d0,d4 beq.b .nocalib move.l d0,d1 lea p_CalibrationArray(a3),a0 move.l a0,d2 move.l #256,d3 call Read cmp.l d0,d3 beq.b .tableloaded .nocalib ; Fill defaults lea p_CalibrationArray(a3),a0 move.w #254-1,d0 .initcalib move.b #$55,(a0)+ dbf d0,.initcalib move.b #$7f,(a0)+ .tableloaded move.l d4,d1 beq.b .nofile call Close .nofile move.l pb_SysLib(a5),a6 move.l #65536*2,d0 move.l #MEMF_PUBLIC,d1 call AllocVec move.l d0,p_CalibrationTable(a3) beq.b .notable move.l d0,a0 ;table lea p_CalibrationArray(a3),a1 bsr _CreateTable .notable * allocate audio.device move.l pb_SysLib(a5),a6 call CreateMsgPort move.l d0,p_audioport(a3) beq.w .error_noport moveq #ioa_SIZEOF,d0 move.l #MEMF_PUBLIC!MEMF_CLEAR,d1 call AllocVec move.l d0,p_audioreq(a3) beq.w .error_noreqmem move.l d0,a0 move.l p_audioport(a3),MN_REPLYPORT(a0) clr.w ioa_AllocKey(a0) move.b #127,LN_PRI(a0) ;steal it! lea .audiochannelarray(pc),a1 move.l a1,ioa_Data(a0) move.l #1,ioa_Length(a0) lea .audioname(pc),a0 moveq #0,d0 move.l p_audioreq(a3),a1 moveq #0,d1 call OpenDevice move.l d0,p_audiodev(a3) bne.w .error_noaudiodev ;somebody already owns the hardware (could be we!) move.l p_audioreq(a3),a1 move.w #CMD_RESET,IO_COMMAND(a1) bsr BeginIO ;clear attach, stop sound. move.l p_audioport(a3),a0 call WaitPort move.l p_audioport(a3),a0 call GetMsg ; move.w #DMAF_AUDIO,DMACON+CUSTOM ;turn audio off ; move.w #$00ff,ADKCON+CUSTOM ;no modulation * allocate interrupts moveq #IS_SIZE,d0 move.l #MEMF_PUBLIC!MEMF_CLEAR,d1 call AllocVec move.l d0,p_Interrupt(a3) beq.w .error_nointmem move.l d0,a0 move.b #NT_INTERRUPT,LN_TYPE(a0) lea LibName(pc),a1 move.l a1,LN_NAME(a0) lea AudioInterrupt(pc),a1 move.l a1,IS_CODE(a0) lea p_AudIntData(a3),a1 move.l a1,IS_DATA(a0) moveq #IS_SIZE,d0 move.l #MEMF_PUBLIC!MEMF_CLEAR,d1 call AllocVec move.l d0,p_NoInterrupt(a3) beq.b .error_nointmem move.l d0,a0 move.b #NT_INTERRUPT,LN_TYPE(a0) lea LibName(pc),a1 move.l a1,LN_NAME(a0) lea Interrupt_Dummy(pc),a1 move.l a1,IS_CODE(a0) clr.l IS_DATA(a0) exg.l a5,a6 call AHIsub_Disable exg.l a5,a6 move.l p_NoInterrupt(a3),a1 moveq #INTB_AUD0,d0 call SetIntVector moveq #IS_SIZE,d0 move.l #MEMF_PUBLIC!MEMF_CLEAR,d1 call AllocVec move.l d0,p_SoftInt(a3) beq.b .error_nointmem move.l d0,a0 move.b #NT_INTERRUPT,LN_TYPE(a0) lea LibName(pc),a1 move.l a1,LN_NAME(a0) lea SoftInt_Dummy(pc),a1 move.l a1,IS_CODE(a0) clr.l IS_DATA(a0) move.l ahiac_MixFreq(a2),d1 bsr calcperiod move.l d0,ahiac_MixFreq(a2) ;store actual freq move.w d1,p_AudPer(a3) moveq #AHISF_KNOWSTEREO|AHISF_MIXING|AHISF_TIMING,d0 .exit popm d2-d7/a2-a6 rts .error_nointmem .error_noaudiodev .error_noreqmem .error_noport .error_nopaula moveq #AHISF_ERROR,d0 bra.b .exit .audiochannelarray dc.b 1+2+4+8 .audioname AUDIONAME .calibname dc.b "ENV:CyberSound/SoundDrivers/14Bit_Calibration",0 ;in: * d1 MixFreq * a5 paulaBase ;out: * d0 New MixFreq * d1.w Period ;description: * Calculate and return the best period and the actual frequency. calcperiod: pushm d2-d4 move.l #PALFREQ,d2 ;PAL move.l pb_GfxLib(a5),a0 move.w gb_DisplayFlags(a0),d0 btst #REALLY_PALn,d0 bne.b .1 move.l #NTSCFREQ,d2 ;NTSC .1 cmp.l #MINFREQ,d1 bhi.b .2 move.l #MINFREQ,d1 .2 move.l d2,d0 move.l d1,d3 move.l pb_UtilLib(a5),a0 jsr _LVOUDivMod32(a0) lsl.l #1,d1 cmp.l d3,d1 bmi.b .3 addq.l #1,d0 .3 move.w d0,d4 moveq #0,d1 move.w d0,d1 move.l d2,d0 move.l d1,d3 jsr _LVOUDivMod32(a0) lsl.l #1,d1 cmp.l d3,d1 bmi.b .4 addq.l #1,d0 .4 move.w d4,d1 popm d2-d4 rts * _CreateTable directly stolen from Christian Buchner's CyberSound * audio sub system (with permission). * _CreateTable ************************************************************** ; Parameters ; a0 = Table address ; (MUST have enough space for 65536 UWORDS) ; a1 = Additive Array ; 256 UBYTEs ; ; the table is organized as follows: ; 32768 UWORDS positive range, ascending order ; 32768 UWORDS negative range, ascending order ; access: (a0,d0.l*2) ; where d0.w is signed word sample data ; and the upper word of d0.l is *cleared!* _CreateTable movem.l a2/d2-d6,-(sp) lea 128(a1),a2 move.l a2,a1 ; count the number of steps moveq #128-1,d0 ; in the positive range moveq #0,d5 .countpositive move.b (a1)+,d1 ext.w d1 ext.l d1 add.l d1,d5 dbra d0,.countpositive ; d5=number of steps move.l #32768,d6 ; reset stretch counter move.l a2,a1 ; middle value in calibdata move.w #32768-1,d0 ; number of positive values -1 moveq #0,d1 ; HI value moveq #0,d2 ; LO value moveq #0,d3 ; counter .fetchnext2 move.b (a1)+,d4 ; add calibtable to counter ext.w d4 add.w d4,d3 .outerloop2 tst.w d3 bgt.s .positive2 .negative2 addq.w #1,d1 ; increment HI value sub.w d4,d2 ; reset LO value bra.s .fetchnext2 .positive2 move.b d1,(a0)+ ; store HI and LO value move.b d2,(a0)+ sub.l d5,d6 ; stretch the table bpl.s .repeat2 ; to 32768 entries add.l #32768,d6 addq.w #1,d2 ; increment LO value subq.w #1,d3 ; decrement counter .repeat2 dbra d0,.outerloop2 move.l a2,a1 ; count the number of steps moveq #128-1,d0 ; in the negative range moveq #0,d5 .countnegative move.b -(a1),d1 ext.w d1 ext.l d1 add.l d1,d5 dbra d0,.countnegative ; d5=number of steps move.l #32768,d6 ; reset stretch counter add.l #2*32768,a0 ; place at the end of the table move.l a2,a1 ; middle value in calibdata move.w #32768-1,d0 ; number of negative values -1 moveq #-1,d1 ; HI value moveq #-1,d2 ; LO value moveq #0,d3 ; counter .fetchnext1 move.b -(a1),d4 ; add calibtable to counter ext.w d4 add.w d4,d3 add.w d4,d2 ; maximize LO value .outerloop1 tst.w d3 bgt.s .positive1 .negative1 subq.w #1,d1 bra.s .fetchnext1 .positive1 move.b d2,-(a0) ; store LO and HI value move.b d1,-(a0) sub.l d5,d6 ; stretch the table bpl.s .repeat1 ; to 32768 entries add.l #32768,d6 subq.w #1,d2 ; decrement lo value subq.w #1,d3 ; decrement counter .repeat1 dbra d0,.outerloop1 movem.l (sp)+,a2/d2-d6 rts ******* .audio/AHIsub_FreeAudio ************************************* * * NAME * AHIsub_FreeAudio -- Deallocates the audio hardware. * * SYNOPSIS * AHIsub_FreeAudio( audioctrl ); * A2 * * void AHIsub_FreeAudio( struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * Deallocate the audio hardware and other resources allocated in * AHIsub_AllocAudio(). * * INPUTS * audioctrl - pointer to an AHIAudioCtrl structure. * * NOTES * It must be safe to call this routine even if AHIsub_AllocAudio() * was never called, failed or called more than once. * * SEE ALSO * AHIsub_AllocAudio() * ***************************************************************************** * * AHIsub_FreeAudio: pushm d2-d7/a2-a6 move.l a6,a5 move.l pb_SysLib(a5),a6 move.l ahiac_DriverData(a2),d0 beq.w .nopaula move.l d0,a3 move.l p_CalibrationTable(a3),d0 beq.b .notable move.l d0,a1 clr.l p_CalibrationTable(a3) call FreeVec .notable move.l p_Interrupt(a3),d0 beq.b .nointstruct move.l d0,a1 clr.l p_Interrupt(a3) call FreeVec .nointstruct move.l p_NoInterrupt(a3),d0 beq.b .nointstruct2 move.l d0,a1 clr.l p_NoInterrupt(a3) call FreeVec .nointstruct2 move.l p_SoftInt(a3),d0 beq.b .nosoftintstruct move.l d0,a1 clr.l p_SoftInt(a3) call FreeVec .nosoftintstruct tst.l p_audiodev(a3) bne.b .noaudiodev move.l p_audioreq(a3),a1 move.w #CMD_RESET,IO_COMMAND(a1) ;Restore interrupts bsr BeginIO move.l p_audioport(a3),a0 call WaitPort move.l p_audioport(a3),a0 call GetMsg move.l p_audioreq(a3),a1 subq.l #1,p_audiodev(a3) call CloseDevice .noaudiodev move.l p_audioreq(a3),d0 beq.b .noaudioreq move.l d0,a1 clr.l p_audioreq(a3) call FreeVec .noaudioreq move.l p_audioport(a3),d0 beq.b .noaudioport move.l d0,a0 clr.l p_audioport(a3) call DeleteMsgPort .noaudioport move.l a3,a1 clr.l ahiac_DriverData(a2) call FreeVec .nopaula moveq #0,d0 popm d2-d7/a2-a6 rts ******* .audio/AHIsub_Disable *************************************** * * NAME * AHIsub_Disable -- Temporary turn off audio interrupt/task * * SYNOPSIS * AHIsub_Disable( audioctrl ); * A2 * * void AHIsub_Disable( struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * If you are lazy, then call exec.library/Disable(). * If you are smart, only disable your own interrupt or task. * * INPUTS * audioctrl - pointer to an AHIAudioCtrl structure. * * NOTES * This call should be guaranteed to preserve all registers. * This call nests. * * SEE ALSO * AHIsub_Enable(), exec.library/Disable() * ***************************************************************************** * * AHIsub_Disable: push a3 move.l ahiac_DriverData(a2),a3 addq.w #1,p_DisableCount(a3) AUDINT_OFF pop a3 rts ******* .audio/AHIsub_Enable **************************************** * * NAME * AHIsub_Enable -- Turn on audio interrupt/task * * SYNOPSIS * AHIsub_Enable( audioctrl ); * A2 * * void AHIsub_Enable( struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * If you are lazy, then call exec.library/Enable(). * If you are smart, only enable your own interrupt or task. * * INPUTS * audioctrl - pointer to an AHIAudioCtrl structure. * * NOTES * This call should be guaranteed to preserve all registers. * This call nests. * * SEE ALSO * AHIsub_Disable(), exec.library/Enable() * ***************************************************************************** * * AHIsub_Enable: push a3 move.l ahiac_DriverData(a2),a3 subq.w #1,p_DisableCount(a3) bne.b .exit AUDINT_ON .exit pop a3 rts ******* .audio/AHIsub_Start ***************************************** * * NAME * AHIsub_Start -- Starts playback or recording * * SYNOPSIS * error = AHIsub_Start( flags, audioctrl ); * D0 D0 A2 * * ULONG AHIsub_Start(ULONG, struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * What to do depends what you returned in AHIsub_AllocAudio(). * * * First, assume bit AHISB_PLAY in flags is set. This means that you * should begin playback. * * - AHIsub_AllocAudio() returned AHISF_MIXING|AHISF_TIMING: * * A) Allocate a mixing buffer of ahiac_BuffSize bytes. * B) Create/start an interrupt or task that will do 1-4 over and over * again until AHIsub_Stop() is called. Note that it is not a good * idea to do the actual mixing and conversion in a real hardware * interrupt. Signal a task or create a Software Interrupt to do * the number crunching. * * 1) Call the user Hook ahiac_PlayerFunc with the following parameters: * A0 - (struct Hook *) * A2 - (struct AHIAudioCtrl *) * A1 - Set to NULL. * 2) Call the mixing Hook (ahiac_MixerFunc) with the following * parameters: * A0 - (struct Hook *) - The Hook itself * A2 - (struct AHIAudioCtrl *) * A1 - (WORD *[]) - The mixing buffer. * Note that ahiac_MixerFunc preserves ALL registers. * The user Hook ahiac_SoundFunc will be called by the mixing * routine when a sample have been processed, so you don't have to * worry about that. * How the buffer will be filled is indicated by ahiac_Flags. * It is allways filled with signed 16-bit (32 bit if AHIDBB_HIFI in * in ahiac_Flags is set) words, even if playback is 8 bit. If * AHIDBB_STEREO is set (in ahiac_Flags), data for left and right * channel are interleved: * 1st sample left channel, * 1st sample right channel, * 2nd sample left channel, * ..., * ahiac_BuffSamples:th sample left channel, * ahiac_BuffSamples:th sample right channel. * If AHIDBB_STEREO is cleared, the mono data is stored: * 1st sample, * 2nd sample, * ..., * ahiac_BuffSamples:th sample. * Note that neither AHIDBB_STEREO nor AHIDBB_HIFI will be set if * you didn't report that you understand these flags when * AHI_AllocAudio() was called. * 3) Convert the buffer if needed and feed it to the audio hardware. * Note that you may have to clear CPU caches if you are using DMA * to play the buffer, and the buffer is not allocated in non- * cachable RAM. * 4) Wait until the whole buffer has been played, then repeat. * * Use double buffering if possible! * * You may DECREASE ahiac_BuffSamples slightly, for example to force an * even number of samples to be mixed. By doing this you will make * ahiac_PlayerFunc to be called at wrong frequency so be careful! * Even if ahiac_BuffSamples is defined ULONG, it will never be greater * than 65535. * * ahiac_BuffSize is the largest size of the mixing buffer that will be * needed until AHIsub_Stop() is called. * * ahiac_MaxBuffSamples is the maximum number of samples that will be * mixed (until AHIsub_Stop() is called). You can use this value if you * need to allocate DMA buffers. * * ahiac_MinBuffSamples is the minimum number of samples that will be * mixed. Most drivers will ignore it. * * * - If AHIsub_AllocAudio() returned AHISF_MIXING, do as described above, * except calling ahiac_PlayerFunc. ahiac_PlayerFunc should be called * ahiac_PlayerFreq times per second, clocked by timers on your sound * card or by using 'realtime.library'. No other Amiga resources may * be used for timing (like direct CIA timers). * ahiac_MinBuffSamples and ahiac_MaxBuffSamples are undefined if * AHIsub_AllocAudio() returned AHISF_MIXING (AHISB_TIMING bit not set). * * - If AHIsub_AllocAudio() returned NULL, start playback. Don't forget to * call ahiac_PlayerFunc ahiac_PlayerFreq times per second. Only your * own timing hardware or 'realtime.library' may be used. * Note that ahiac_MixFreq, ahiac_MixerFunc, ahiac_BuffSamples, * ahiac_MinBuffSamples, ahiac_MaxBuffSamples and ahiac_BuffSize are * undefined if AHIsub_AllocAudio() returned NULL. * * * * Second, assume bit AHISB_RECORD in flags is set. This means that you * should start to sample. Create a interrupt or task that does the * following: * * Allocate a buffer (you chose size, but try to keep it reasonable * small to avoid delays), and fill it with the sampled data. The format * should always be AHIST_S16S (even with 8 bit mono samplers), which * means: * 1st sample left channel, * 1st sample right channel (same as prev. if mono), * 2nd sample left channel, * ... etc. * Each sample is a signed word (WORD). The sample rate should be equal * to the mixing rate. * * Call the ahiac_SamplerFunc Hook with the following parameters: * A0 - (struct Hook *) - The Hook itself * A2 - (struct AHIAudioCtrl *) * A1 - (struct AHIRecordMessage *) * The message should be filled as follows: * ahism_Buffer - A pointer to the filled buffer. * ahism_Samples - How many sample frames stored. * ahism_Type - Set to AHIST_S16S. * You must not destroy the buffer until next time the Hook is called. * * Repeat until AHIsub_Stop() is called. * * INPUTS * flags - See . * audioctrl - pointer to an AHIAudioCtrl structure. * * RESULT * Returns AHIE_OK if successful, else an error code as defined * in . AHIsub_Stop() will always be called, even * if this call failed. * * NOTES * * SEE ALSO * AHIsub_Update(), AHIsub_Stop() * ***************************************************************************** * * * AHIsub_Start: pushm d2-d7/a2-a6 btst #AHISB_PLAY,d0 beq.w .return call AHIsub_Update ;fill variables lea CUSTOM,a4 move.l a6,a5 move.l pb_SysLib(a5),a6 move.l ahiac_DriverData(a2),a3 move.l p_SoftInt(a3),a0 ;initXXX will fill IS_CODE lea p_SoftIntData(a3),a1 move.l a1,IS_DATA(a0) pea .1(pc) move.b p_Flags(a3),d0 and.b #PF_STEREO!PF_14BIT,d0 beq.w init8bitM cmp.b #PF_STEREO,d0 beq.w init8bitS cmp.b #PF_14BIT,d0 beq.w init14bitM bra.w init14bitS .1 tst.l d0 bne.b .exit move.l ahiac_BuffSize(a2),d0 move.l #MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_Mixbuffer(a3) beq.b .error_nomixmem move.l ahiac_MixFreq(a2),d1 bsr calcperiod move.w d1,p_AudPer(a3) ; bset.b #PB_INT,p_Flags(a3) ; move.w DMACONR(a4),d0 ; and.w #DMAF_AUDIO,d0 ; or.w #DMAF_SETCLR,d0 ; move.w d0,p_OldDMACON(a3) ; move.w INTENAR(a4),d0 ; and.w #INTF_AUDIO,d0 ; or.w #INTF_SETCLR,d0 ; move.w d0,p_OldINTENA(a3) move.l p_Interrupt(a3),a1 moveq #INTB_AUD0,d0 call SetIntVector clr.w p_DisableCount(a3) AUDINT_ON move.w #INTF_SETCLR!INTF_AUD0,INTREQ(a4) ;start .return moveq #AHIE_OK,d0 .exit popm d2-d7/a2-a6 rts .error_nomixmem moveq #AHIE_NOMEM,d0 bra.b .exit init_error_nodmamem moveq #AHIE_NOMEM,d0 rts ;in: * a2 AudioCtrl * a3 paula * a4 CUSTOM * a5 paulaBase * a6 ExecBase init8bitM: move.l p_SoftInt(a3),a0 lea SoftInt_8bitM(pc),a1 move.l a1,IS_CODE(a0) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 8 bit samples move.l d0,d2 ;d2 = channel size lsl.l #1,d0 ;Double buffer move.l #MEMF_CHIP!MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_DMAbuffer(a3) beq.b init_error_nodmamem move.l d0,p_AudPtr1A(a3) move.l d0,p_AudPtr2A(a3) add.l d2,d0 move.l d0,p_AudPtr1B(a3) move.l d0,p_AudPtr2B(a3) move.w #64,AUD0VOL(a4) move.w #64,AUD1VOL(a4) moveq #0,d0 rts ;in: * a2 AudioCtrl * a3 paula * a5 paulaBase * a6 ExecBase init8bitS: move.l p_SoftInt(a3),a0 lea SoftInt_8bitS(pc),a1 move.l a1,IS_CODE(a0) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 8 bit samples move.l d0,d2 ;d2 = channel size lsl.l #2,d0 ;Double buffer + Stereo move.l #MEMF_CHIP!MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_DMAbuffer(a3) beq.w init_error_nodmamem move.l d0,p_AudPtr1A(a3) add.l d2,d0 move.l d0,p_AudPtr2A(a3) add.l d2,d0 move.l d0,p_AudPtr1B(a3) add.l d2,d0 move.l d0,p_AudPtr2B(a3) move.w #64,AUD0VOL(a4) move.w #64,AUD1VOL(a4) moveq #0,d0 rts ;in: * a2 AudioCtrl * a3 paula * a5 paulaBase * a6 ExecBase init14bitM: move.l p_SoftInt(a3),a0 lea SoftInt_14bitM(pc),a1 tst.l p_CalibrationTable(a3) beq.b .nocalib lea SoftInt_14CbitM(pc),a1 .nocalib move.l a1,IS_CODE(a0) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 16 bit samples move.l d0,d2 ;d2 = channel size lsl.l #2,d0 ;Double buffer + 2×8 bit move.l #MEMF_CHIP!MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_DMAbuffer(a3) beq.w init_error_nodmamem move.l d0,p_AudPtr1A(a3) move.l d0,p_AudPtr2A(a3) add.l d2,d0 move.l d0,p_AudPtr4A(a3) move.l d0,p_AudPtr3A(a3) add.l d2,d0 move.l d0,p_AudPtr1B(a3) move.l d0,p_AudPtr2B(a3) add.l d2,d0 move.l d0,p_AudPtr4B(a3) move.l d0,p_AudPtr3B(a3) move.w #64,AUD0VOL(a4) move.w #64,AUD1VOL(a4) move.w #1,AUD2VOL(a4) move.w #1,AUD3VOL(a4) moveq #0,d0 rts ;in: * a2 AudioCtrl * a3 paula * a5 paulaBase * a6 ExecBase init14bitS: move.l p_SoftInt(a3),a0 lea SoftInt_14bitS(pc),a1 tst.l p_CalibrationTable(a3) beq.b .nocalib lea SoftInt_14CbitS(pc),a1 .nocalib move.l a1,IS_CODE(a0) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 16 bit samples move.l d0,d2 ;d2 = channel size lsl.l #3,d0 ;Double buffer + 2×8 bit + Stereo move.l #MEMF_CHIP!MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_DMAbuffer(a3) beq.w init_error_nodmamem move.l d0,p_AudPtr1A(a3) add.l d2,d0 move.l d0,p_AudPtr2A(a3) add.l d2,d0 move.l d0,p_AudPtr3A(a3) add.l d2,d0 move.l d0,p_AudPtr4A(a3) add.l d2,d0 move.l d0,p_AudPtr1B(a3) add.l d2,d0 move.l d0,p_AudPtr2B(a3) add.l d2,d0 move.l d0,p_AudPtr3B(a3) add.l d2,d0 move.l d0,p_AudPtr4B(a3) move.w #64,AUD0VOL(a4) move.w #64,AUD1VOL(a4) move.w #1,AUD2VOL(a4) move.w #1,AUD3VOL(a4) moveq #0,d0 rts ******* .audio/AHIsub_Update **************************************** * * NAME * AHIsub_Update -- Update some variables * * SYNOPSIS * AHIsub_Update( flags, audioctrl ); * D0 A2 * * void AHIsub_Update(ULONG, struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * All you have to do is to update some variables: * Mixing & timing: ahiac_PlayerFunc, ahiac_MixerFunc, * ahiac_BuffSamples (and perhaps ahiac_PlayerFreq if you use it). * Mixing only: ahiac_PlayerFunc, ahiac_MixerFunc and ahiac_PlayerFreq. * Nothing: ahiac_PlayerFunc and ahiac_PlayerFreq. * * INPUTS * flags - Currently no flags defined. * audioctrl - pointer to an AHIAudioCtrl structure. * * RESULT * * NOTES * This call must be safe from interrupts. * * SEE ALSO * AHIsub_Start() * ***************************************************************************** * * * AHIsub_Update: pushm d2-d7/a2-a6 call AHIsub_Disable ;make sure we don't get an interrupt ;while updating our local variables move.l ahiac_DriverData(a2),a3 move.l ahiac_PlayerFunc(a2),a0 move.l a0,p_PlayerHook(a3) move.l h_Entry(a0),p_PlayerEntry(a3) move.l ahiac_BuffSamples(a2),d0 lsr.l #1,d0 ;length in words, force even # of samples move.w d0,p_AudLen(a3) subq.l #1,d0 move.l d0,p_LoopTimes(a3) ;See softints. (Unrolled) move.l ahiac_MixerFunc(a2),a0 move.l a0,p_MixHook(a3) move.l h_Entry(a0),p_MixEntry(a3) call AHIsub_Enable moveq #0,d0 popm d2-d7/a2-a6 rts ******* .audio/AHIsub_Stop ****************************************** * * NAME * AHIsub_Stop -- Stops playback. * * SYNOPSIS * AHIsub_Stop( flags, audioctrl ); * D0 A2 * * void AHIsub_Stop( ULONG, struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * Stop playback and/or recording, remove all resources allocated by * AHIsub_Start(). * * INPUTS * flags - See . * audioctrl - pointer to an AHIAudioCtrl structure. * * NOTES * It must be safe to call this routine even if AHIsub_Start() was never * called, failed or called more than once. * * SEE ALSO * AHIsub_Start() * ***************************************************************************** * * AHIsub_Stop: pushm d2-d7/a2-a6 btst #AHISB_PLAY,d0 beq.b .return move.l a6,a5 move.l pb_SysLib(a5),a6 move.l ahiac_DriverData(a2),a3 lea CUSTOM,a4 moveq #0,d0 move.w d0,AUD0VOL(a4) move.w d0,AUD1VOL(a4) move.w d0,AUD2VOL(a4) move.w d0,AUD3VOL(a4) move.w #DMAF_AUDIO,DMACON(a4) ;disable audio DMA ; move.w p_OldDMACON(a3),DMACON(a4) ;restore audio DMA ; bclr.b #PB_INT,p_Flags(a3) ; beq.b .noint AUDINT_OFF move.w #INTF_AUDIO,INTREQ(a4) ;Clear any waiting interrupts move.l p_NoInterrupt(a3),a1 moveq #INTB_AUD0,d0 call SetIntVector ; move.w p_OldINTENA(a3),INTENA(a4) ;restore interrupts .noint move.l p_DMAbuffer(a3),d0 beq.b .nodmamem move.l d0,a1 clr.l p_DMAbuffer(a3) call FreeVec .nodmamem move.l p_Mixbuffer(a3),d0 beq.b .nomixmem move.l d0,a1 clr.l p_Mixbuffer(a3) call FreeVec .nomixmem .return moveq #0,d0 popm d2-d7/a2-a6 rts ******* .audio/AHIsub_#? ******************************************** * * NAME * AHIsub_SetEffect -- Set effect. * AHIsub_SetFreq -- Set frequency. * AHIsub_SetSound -- Set sound. * AHIsub_SetVol -- Set volume and stereo panning. * AHIsub_LoadSound -- Prepare a sound for playback. * AHIsub_UnLoadSound -- Discard a sound. * * SYNOPSIS * See functions in 'ahi.library'. * * DO-IT-YOURSELF * If AHIsub_AllocAudio() did not return with bit AHISB_MIXING set, * all user calls to these function will be routed to the driver. * * If AHIsub_AllocAudio() did return with bit AHISB_MIXING set, the * calls will first be routed to the driver, and only handled by * 'ahi.library' if the driver returned AHIS_UNKNOWN. This way it is * possible to add, for example, a master volume effect that takes * advantage of hardware volume control, and falls back to digital * volume control if that failed. * * For what each funtion does, see the autodocs for 'ahi.library'. * * If your hardware supports it, you are encouraged to handle all * AHIST_INPUT sounds yourself. The volume is calculated as * SUM(Volume)/Channels (linear) of all currently playing sounds of this * type. You need to intercept AHIsub_SetFreq, AHIsub_SetSound, * AHIsub_SetVol and AHIsub_LoadSound for this feature. * * INPUTS * See functions in 'ahi.library'. * * NOTES * See functions in 'ahi.library'. * * SEE ALSO * ahi.library/AHI_SetEffect(), ahi.library/AHI_SetFreq(), * ahi.library/AHI_SetSound(), ahi.library/AHI_SetVol(), * ahi.library/AHI_LoadSound(), ahi.library/AHI_UnLoadSound() * * ***************************************************************************** * * AHIsub_SetVol: AHIsub_SetFreq: AHIsub_SetSound: AHIsub_SetEffect: AHIsub_LoadSound: AHIsub_UnLoadSound: moveq #AHIS_UNKNOWN,d0 rts ******* .audio/AHIsub_GetAttr *************************************** * * NAME * AHIsub_GetAttr -- Returns information about audio modes or driver * * SYNOPSIS * AHIsub_GetAttr( attribute, argument, default, taglist, audioctrl ); * D0 D0 D1 D2 A1 a2 * * LONG AHIsub_GetAttr( ULONG, LONG, LONG, struct TagItem *, * struct AHIAudioCtrl * ); * * DO-IT-YOURSELF * Return the attribute based on a tag list and an AHIAudioCtrl * structure, which are the same that will be passed to * AHIsub_AllocAudio() by 'ahi.library'. If the attribute is * unknown to you, return the default. * * INPUTS * attribute - Is really a Tag and can be one of the following: * AHIDB_Bits - Return how many output bits the tag list will * result in. * AHIDB_MaxChannels - Return the resulting number of channels. * AHIDB_Frequencies - Return how many mixing/sampling frequencies * you support * AHIDB_Frequency - Return the argument:th frequency * Example: You support 3 frequencies 32, 44.1 and 48 kHz. * If argument is 1, return 44100. * AHIDB_Index - Return the index which gives the frequency closest * to argument. * Example: You support 3 frequencies 32, 44.1 and 48 kHz. * If argument is 40000, return 1 (=> 44100)- * AHIDB_Author - Return pointer to name of driver author: * "Martin 'Leviticus' Blom" * AHIDB_Copyright - Return pointer to copyright notice, including * the '©' character: "© 1996 Martin Blom" or "Public Domain" * AHIDB_Version - Return pointer version string, normal Amiga * format: "paula 1.5 (18.2.96)\r\n" * AHIDB_Annotation - Return pointer to an annotation string, which * can be several lines. * argument - extra info for some attributes. * default - What you should return for unknown attributes. * taglist - Pointer to a tag list that eventually will be fed to * AHIsub_AllocAudio(), or NULL. * audioctrl - Pointer to an AHIAudioCtrl structure that eventually * will be fed to AHIsub_AllocAudio(), or NULL. * * NOTES * * SEE ALSO * AHIsub_AllocAudio() * ***************************************************************************** * * AHIsub_GetAttr: pushm d2/a5-a6 move.l a6,a5 tst.l a1 beq.b .no14bit ;no tag list! cmp.l #AHIDB_Bits,d0 bne.b .not_bits move.l pb_UtilLib(a5),a6 move.l #AHIDB_Paula14Bit,d0 moveq #FALSE,d1 move.l a1,a0 call GetTagData tst.l d0 beq.b .no14bit moveq #14,d0 skipw .no14bit moveq #8,d0 bra.w .exit .not_bits cmp.l #AHIDB_Frequencies,d0 bne.b .not_freqs move.l #MAXPER-MINPER+1,d0 bra.w .exit .not_freqs cmp.l #AHIDB_Frequency,d0 bne.b .not_freq sub.w #MAXPER,d1 neg.w d1 move.l #PALFREQ,d2 ;PAL move.l pb_GfxLib(a5),a0 move.w gb_DisplayFlags(a0),d0 btst #REALLY_PALn,d0 bne.b .1 move.l #NTSCFREQ,d2 ;NTSC .1 divu.w d1,d2 moveq #0,d0 move.w d2,d0 bra.b .exit .not_freq cmp.l #AHIDB_Index,d0 bne.b .not_index bsr calcperiod sub.w #MAXPER,d1 neg.w d1 moveq #0,d0 move.w d1,d0 bra.b .exit .not_index cmp.l #AHIDB_Author,d0 bne.b .not_author lea .author(pc),a0 move.l a0,d0 bra.b .exit .not_author cmp.l #AHIDB_Copyright,d0 bne.b .not_copyright lea .copyright(pc),a0 move.l a0,d0 bra.b .exit .not_copyright cmp.l #AHIDB_Version,d0 bne.b .not_version lea IDString(pc),a0 move.l a0,d0 bra.b .exit .not_version cmp.l #AHIDB_Annotation,d0 bne.b .not_anno lea .anno(pc),a0 move.l a0,d0 bra.b .exit .not_anno * Unknown attribute, return default. move.l d2,d0 .exit popm d2/a5-a6 rts .author dc.b "Martin 'Leviticus' Blom",0 .copyright dc.b "Public Domain",0 .anno dc.b "14 bit routines by Christian Buchner.",0 even ;---------------------------------------------------------------------------- Interrupt_Dummy: move.w d1,INTREQ(a0) SoftInt_Dummy: rts ;in: * d0 scratch * d1 INTENAR && INTREQR * a0 CUSTOM * a1 &(paulaBase->p_AudIntData) * a5 &AudioInterrupt * a6 ExecBase AudioInterrupt: move.w d1,INTREQ(a0) movem.l (a1)+,d0/d1 ;p_SoftInt,p_AudLen/p_AudPer move.l d1,AUD0LEN(a0) move.l d1,AUD1LEN(a0) move.l d1,AUD2LEN(a0) move.l d1,AUD3LEN(a0) not.l (a1)+ ;double buffering lea p_AudPtrs-p_AudPtr1A(a1),a5 beq.b .1 add.w #4*4,a1 .1 move.l a1,(a5) move.l (a1)+,AUD0LC(a0) move.l (a1)+,AUD1LC(a0) move.l (a1)+,AUD2LC(a0) move.l (a1)+,AUD3LC(a0) move.l d0,a1 ;p_SoftInt jmp _LVOCause(a6) ;start SoftInt ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_SoftIntData) * a5 scratch SoftInt_8bitM: pushm a2/a3 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3 jsr (a3) ;call Player Hook movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer move.l (a5),a0 .loop move.b (a1),(a0)+ addq.l #2,a1 move.b (a1),(a0)+ addq.l #2,a1 dbf d0,.loop popm a2/a3 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_SoftIntData) * a5 scratch SoftInt_8bitS: pushm a2/a3 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3 jsr (a3) ;call Player Hook movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer movem.l (a5),a0/a2 .loop REPT 2 ;left move.b (a1),(a0)+ ;right move.b 2(a1),(a2)+ addq.l #4,a1 ENDR dbf d0,.loop popm a2/a3 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_SoftIntData) * a5 scratch SoftInt_14bitM: pushm a2/a3 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3 jsr (a3) ;call Player Hook movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer movem.l (a5),a0/a2/a3/a5 ;a2/a3 unused .loop REPT 2 move.b (a1)+,(a0)+ move.b (a1)+,d1 asr.b #2,d1 move.b d1,(a5)+ ENDR dbf d0,.loop popm a2/a3 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1|DMAF_AUD2|DMAF_AUD3,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_SoftIntData) * a5 scratch SoftInt_14CbitM: pushm a2/a3/a4 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3 jsr (a3) ;call Player Hook movem.l (a5),d0/a0/a1/a2/a3/a4/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer, using the table movem.l (a4),a0/a2/a3/a4 ;a2/a3 unused moveq #0,d1 .loop REPT 2 IFNE MC020p move.w (a1)+,d1 ;fetch 16 bit sample move.w (a5,d1.l*2),d1 ELSE moveq #0,d1 move.w (a1)+,d1 ;fetch 16 bit sample add.w d1,d1 move.w (a5,d1.l),d1 ENDC move.b d1,(a4)+ ;low byte lsr.w #8,d1 move.b d1,(a0)+ ;high byte ENDR dbf d0,.loop popm a2/a3/a4 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1|DMAF_AUD2|DMAF_AUD3,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_SoftIntData) * a5 scratch SoftInt_14bitS: pushm a2/a3 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3 jsr (a3) ;call Player Hook movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer * TODO: Use a table! movem.l (a5),a0/a2/a3/a5 .loop REPT 2 ;left move.b (a1)+,(a0)+ move.b (a1)+,d1 asr.b #2,d1 move.b d1,(a5)+ ;right move.b (a1)+,(a2)+ move.b (a1)+,d1 asr.b #2,d1 move.b d1,(a3)+ ENDR dbf d0,.loop popm a2/a3 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1|DMAF_AUD2|DMAF_AUD3,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_SoftIntData) * a5 scratch SoftInt_14CbitS: pushm a2/a3/a4 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3 jsr (a3) ;call Player Hook movem.l (a5),d0/a0/a1/a2/a3/a4/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer, using the table movem.l (a4),a0/a2/a3/a4 moveq #0,d1 .loop REPT 2 ;left IFNE MC020p move.w (a1)+,d1 ;fetch 16 bit sample move.w (a5,d1.l*2),d1 ELSE moveq #0,d1 move.w (a1)+,d1 ;fetch 16 bit sample add.w d1,d1 move.w (a5,d1.l),d1 ENDC move.b d1,(a4)+ ;low byte lsr.w #8,d1 move.b d1,(a0)+ ;high byte ;right IFNE MC020p move.w (a1)+,d1 ;fetch 16 bit sample move.w (a5,d1.l*2),d1 ELSE moveq #0,d1 move.w (a1)+,d1 ;fetch 16 bit sample add.w d1,d1 move.w (a5,d1.l),d1 ENDC move.b d1,(a3)+ ;low byte lsr.w #8,d1 move.b d1,(a2)+ ;high byte ENDR dbf d0,.loop popm a2/a3/a4 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1|DMAF_AUD2|DMAF_AUD3,CUSTOM+DMACON rts EndCode: