*** ScR *** *** NOTES *** * There is still room for improvements in the softint's. For example, * try if reducing chip memory accesses increases speed. Not high priority, though. * This driver should probably have a DMA play mode for 1-4 stereo channels. * * The sampler routines are just for fun. Since interrupts must not be disabled, * there are lots of clicks. But still, I have tried to sample at 22kHz while down- * loading files from a BBS on my A4000/040, and it worked without any serial errors. ;------------ VERSION EQU 4 REVISION EQU 2 DATE MACRO dc.b "17.4.97" ENDM VERS MACRO dc.b "paula 4.2" 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 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 lvo/cardres_lib.i include devices/ahi.i include libraries/ahi_sub.i include lvo/ahi_sub_lib.i include resources/misc.i include resources/card.i include macros.i _ciaa EQU $bfe001 _ciab EQU $bfd000 PALFREQ EQU 3546895 NTSCFREQ EQU 3579545 MINPER EQU 62 RECORDSAMPLES EQU 1024 IFD BARFLY IFD MC020 MC68020 ENDC ELSE IFD MC020 auto wo AHI:user/devs/ahi/paula.audio\ ELSE auto wo AHI:user/devs/ahi/paula.audio.000\ ENDC ENDC * BARFLY * paula.audio extra tags AHIDB_Paula14Bit EQU AHIDB_UserBase+0 * Boolean AHIDB_PaulaTable EQU AHIDB_UserBase+1 * Boolean * 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 APTR pb_MiscResource APTR pb_CardResource LABEL paulaBase_SIZEOF * paula (private) ahiac_DriverData points to this structure. STRUCTURE paula,0 UBYTE p_Flags UBYTE p_Parallel ;TRUE if pport allocated UWORD p_DisableCount APTR p_audioport APTR p_audioreq ULONG p_audiodev APTR p_DMAbuffer APTR p_ParBitsUser APTR p_ParPortUser APTR p_CardHandle STRUCT p_PlayInt,IS_SIZE STRUCT p_PlaySoftInt,IS_SIZE STRUCT p_RecInt,IS_SIZE STRUCT p_RecSoftInt,IS_SIZE UWORD p_MonitorVolume UWORD p_Input *** IS_DATA for Play Interrupt (do not change order!) LABEL p_PlayIntData APTR p_PlaySoftIntPtr 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 Play Software Interrupt (do not change order!) LABEL p_PlaySoftIntData APTR p_PlayerHook ULONG p_Reserved APTR p_AudioCtrlP FPTR p_PlayerEntry ;p_PlayerHook->h_Entry FPTR p_PreTimer FPTR p_PostTimer 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 *** IS_DATA for Aura Record Interrupt (do not change order!) LABEL p_RecIntDataAura APTR p_AuraAddress *** IS_DATA for Paula Record Interrupt (do not change order!) LABEL p_RecIntData APTR p_RecFillPtr UWORD p_RecFillCount UWORD p_Pad3 APTR p_RecBuffer1 APTR p_RecBuffer2 APTR p_RecSoftIntPtr LABEL p_RecordMessage ULONG p_rmType APTR p_rmBuffer ULONG p_rmLength STRUCT p_CalibrationArray,256 LABEL paula_SIZEOF * p_Flags BITDEF P,14BIT,0 PB_STEREO EQU AHIACB_STEREO ;=2 PF_STEREO EQU AHIACF_STEREO Start: moveq #-1,d0 rts 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 miscName: MISCNAME cardName: dc.b "card.resource",0 envVar: dc.b "AHIpaulaFilterFreq",0 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 AHIsub_HardwareControl 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 .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 .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 .exit .dosOK lea miscName(pc),a1 call OpenResource move.l d0,pb_MiscResource(a5) bne.b .miscOK ALERT AG_OpenRes!AO_MiscRsrc moveq #0,d0 bra .exit .miscOK lea cardName(pc),a1 call OpenResource move.l d0,pb_CardResource(a5) ;Don't fail on error 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 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 ****** [driver].audio/--background-- **************************************** * * OVERVIEW * * AUDIO ID NUMBERS * * Just some notes about selecting ID numbers for different modes: * It is up to the driver programmer to chose which modes should be * available to the user. Take care when selecting. * The upper word is the hardware ID, and can only be allocated by * Martin Blom . The lower word is free, but in * order to allow enhancements, please only use bit 0 to 3 for modes! * Set the remaining bits to zero. * The ID is currently used by AHI_BestAudioIDA() to select a mode. * In order for this to work, you have to follow some rules. * · Modes using tables should have higher IDs than modes that uses more * CPU power. * · Mono modes should have higher IDs than stereo modes * · Stereo modes should have higher IDs than stereo modes with panning. * Use AHI:Developer/Support/ScanAudioModes to have a look at the modes * currently available. * * I do reserve the right to change the rules if I find them incorrect! * ***************************************************************************** * * ****** [driver].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 AHIAudioCtrlDrv * ); * * IMPLEMENTATION * Allocate and initialize the audio hardware. Decide if and how you * wish to use the mixing routines provided by 'ahi.device', by looking * in the AHIAudioCtrlDrv 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 the hardware can't play samples with any length, use this * alternative and provide timing yourself. The buffer must * take less than about 20 ms to play, preferable less than 10! * · 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 * Alternatively, you can use the first method and call the * mixing hook several times in a row to fill up a buffer. * In that case, AHIsub_GetAttr(AHIDB_MaxPlaySamples) should * return the size of the buffer plus AudioCtrl->ahiac_MaxBuffSamples. * If the buffer is so large that it takes more than (approx.) 10 ms to * play it for high sample frequencies, AHIsub_GetAttr(AHIDB_Realtime) * should return FALSE. * 3) Don't use mixing routines: * If your hardware can handle everything without using the CPU to * mix the channels, you tell 'ahi.device' this by not setting * neither the AHISB_MIXING nor the AHISB_TIMING bit. * * If you can handle stereo output from the mixing routines, also set * bit AHISB_KNOWSTEREO. * * If you can handle hifi (32 bit) output from the mixing routines, * set bit AHISB_KNOWHIFI. * * If this driver can be used to record samples, set bit AHISB_CANRECORD, * too (regardless if you use the mixing routines in AHI or not). * * If the sound card has hardware to do DSP effects, you can set the * AHISB_CANPOSTPROCESS bit. The output from the mixing routines will * then be two separate buffers, one wet and one dry. You should then * apply the Fx on the wet buffer, and post-mix the two buffers before * you send the samples to the DAC. (V4) * * INPUTS * tags - pointer to a taglist. * audioctrl - pointer to an AHIAudioCtrlDrv structure. * * TAGS * The tags are from the audio database (AHIDB_#? in ), * NOT the tag list the user called ahi.device/AHI_AllocAudio() with. * * RESULT * Flags, defined in . * * EXAMPLE * * NOTES * You don't have to clean up on failure, AHIsub_FreeAudio() will * always be called. * * BUGS * * SEE ALSO * AHIsub_FreeAudio(), AHIsub_Start() * ***************************************************************************** * * AHIsub_AllocAudio: pushm d2-d7/a2-a6 move.l a6,a5 move.l a1,d3 * Allocate the 'paula' structure (our variables) 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 .error_nopaula move.l d0,a3 * Initialize some fields... move.l #-1,p_audiodev(a3) move.l #-1,p_ParBitsUser(a3) move.l #-1,p_ParPortUser(a3) move.l a2,p_AudioCtrlP(a3) ;player Hook move.l a2,p_AudioCtrlM(a3) ;mixer Hook lea p_PlaySoftInt(a3),a0 move.l a0,p_PlaySoftIntPtr(a3) lea p_RecSoftInt(a3),a0 move.l a0,p_RecSoftIntPtr(a3) move.l #AHIST_S16S,p_rmType(a3) move.l #RECORDSAMPLES,p_rmLength(a3) * Translate tags to flags 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) * Check if a table should be used (14 bit calibration) move.l #AHIDB_PaulaTable,d0 moveq #0,d1 move.l d3,a0 ;tag list call GetTagData tst.l d0 beq .notable * Load 'ENV:CyberSound/SoundDrivers/14Bit_Calibration', allocate * and initialize the table. * FIXIT: The calibration file should move to a special chunk 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 .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 .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.w _CreateTable .notable * allocate audio.device move.l pb_SysLib(a5),a6 call CreateMsgPort move.l d0,p_audioport(a3) beq .error_noport moveq #ioa_SIZEOF,d0 move.l #MEMF_PUBLIC!MEMF_CLEAR,d1 call AllocVec move.l d0,p_audioreq(a3) beq .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 .error_noaudiodev ;somebody already owns the hardware (could be us!) move.l p_audioreq(a3),a1 move.w #CMD_RESET,IO_COMMAND(a1) bsr.w BeginIO ;clear attach, stop sound. move.l p_audioport(a3),a0 call WaitPort move.l p_audioport(a3),a0 call GetMsg move.l pb_DosLib(a5),a6 moveq #1,d1 call Delay * test if mode supports recording btst #PB_14BIT,p_Flags(a3) bne .dontgetsampler ;no record if 14 bit mode * try to allocate parallel port clr.b p_Parallel(a3) move.l pb_MiscResource(a5),a6 moveq #MR_PARALLELBITS,d0 lea IDString(pc),a1 jsr MR_ALLOCMISCRESOURCE(a6) move.l d0,p_ParBitsUser(a3) bne .no_parrallel moveq #MR_PARALLELPORT,d0 lea IDString(pc),a1 jsr MR_ALLOCMISCRESOURCE(a6) move.l d0,p_ParPortUser(a3) bne .no_parrallel move.b #TRUE,p_Parallel(a3) move.b #0,_ciaa+ciaddrb ;make PB0-PB7 inputs .no_parrallel * allocate Aura sampler clr.l p_AuraAddress(a3) move.l pb_CardResource(a5),d0 beq .no_aura move.l d0,a6 call GetCardMap tst.l d0 beq .no_aura move.l d0,a0 move.l cmm_IOMemory(a0),d2 beq .no_aura base exec moveq #CardHandle_SIZEOF,d0 move.l #MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_CardHandle(a3) beq .no_aura move.l pb_CardResource(a5),a6 move.l d0,a1 move.l #IDString,LN_NAME(a1) move.b #CARDF_RESETREMOVE|CARDF_IFAVAILABLE,cah_CardFlags(a1) call OwnCard tst.l d0 bne .no_aura move.l p_CardHandle(a3),a1 call BeginCardAccess tst.l d0 beq .no_aura move.l d2,p_AuraAddress(a3) .no_aura .dontgetsampler * initialize interrupts (Only dummy function pointers at this time) * p_PlayInt (the main playback interrupt) move.b #NT_INTERRUPT,LN_TYPE+p_PlayInt(a3) move.l #LibName,LN_NAME+p_PlayInt(a3) move.l #Interrupt_Dummy,IS_CODE+p_PlayInt(a3) lea p_PlayIntData(a3),a1 move.l a1,IS_DATA+p_PlayInt(a3) * p_PlaySoftInt (caused by p_PlayInt, here are the mixing and conversion done) move.b #NT_INTERRUPT,LN_TYPE+p_PlaySoftInt(a3) move.l #LibName,LN_NAME+p_PlaySoftInt(a3) move.l #SoftInt_Dummy,IS_CODE+p_PlaySoftInt(a3) lea p_PlaySoftIntData(a3),a1 move.l a1,IS_DATA+p_PlaySoftInt(a3) * p_RecInt (the interrupt used for recording) move.b #NT_INTERRUPT,LN_TYPE+p_RecInt(a3) move.l #LibName,LN_NAME+p_RecInt(a3) move.l #Interrupt_Dummy,IS_CODE+p_RecInt(a3) clr.l IS_DATA+p_RecInt(a3) * p_RecSoftInt (caused by p_RecInt when the record buffer has been filled) move.b #32,LN_PRI+p_RecSoftInt(a3) move.b #NT_INTERRUPT,LN_TYPE+p_RecSoftInt(a3) move.l #LibName,LN_NAME+p_RecSoftInt(a3) move.l #RecordSoftInt,IS_CODE+p_RecSoftInt(a3) move.l a3,IS_DATA+p_RecSoftInt(a3) * Make sure no interrupts occur until AHIsub_Start() is called move.w #INTF_AUD0!INTF_AUD1!INTF_AUD2!INTF_AUD3,CUSTOM+INTENA * Update ahiac_MixFreq to what the mixing/sampling frequency really is move.l ahiac_MixFreq(a2),d1 bsr calcperiod move.l d0,ahiac_MixFreq(a2) ;store actual freq * Check the AHIpaulaFilterFreq variable. * If the mixing frequency is higher than this one, set disable the filter, * else enable it. moveq #0,d5 ;Default freq move.l pb_DosLib(a5),a6 subq.l #8,sp ;local label move.w #("0"<<8)|0,(sp) ;Initialize as "0" lea envVar(pc),a0 move.l a0,d1 move.l sp,d2 moveq.l #8,d3 moveq.l #0,d4 call GetVar cmp.l #-1,d0 beq .gotfreq move.l sp,d1 pea.l 0.w move.l sp,d2 call StrToLong move.l (sp)+,d5 .gotfreq addq.l #8,sp ; d5 is now the freq cmp.l ahiac_MixFreq(a2),d5 bls .no_filter bclr #1,$bfe001 ;turn audio filter on bra .filter_set .no_filter bset #1,$bfe001 ;turn audio filter off .filter_set moveq #AHISF_KNOWSTEREO|AHISF_CANRECORD|AHISF_MIXING|AHISF_TIMING,d0 .exit popm d2-d7/a2-a6 rts .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 even .cardhandle: dc.l 0,0 ;ln_Succ, ln_Pred dc.b 0 ;ln_Type dc.b 0 ;ln_Pri dc.l IDString ;ln_Name dc.l 0,0,0 ;cah_CardRemoved, cah_CardInserted, cah_CardStatus dc.b (CARDF_RESETREMOVE|CARDF_IFAVAILABLE) even ;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 move.l d2,d0 move.l d1,d3 move.l pb_UtilLib(a5),a1 jsr _LVOUDivMod32(a1) lsl.l #1,d1 cmp.l d3,d1 bmi.b .3 addq.l #1,d0 .3 move.w d0,d4 * d4 is now period. Check if is it a valid one, depending on current display mode bsr checkvideo moveq #MINPER,d1 tst.l d0 bne.b .5 moveq #MINPER*2,d1 .5 cmp.w d1,d4 bhs.b .6 move.w d1,d4 .6 moveq #0,d1 move.w d4,d1 move.l d2,d0 move.l d1,d3 jsr _LVOUDivMod32(a1) lsl.l #1,d1 cmp.l d3,d1 bmi.b .4 addq.l #1,d0 .4 move.w d4,d1 popm d2-d4 rts ;in: * a5 paulaBase ;out: * d0 TRUE if current mode is double ;description: * Checks if the current screen mode is doublescan. * This routine is *pretty* ugly, but it does get the job * done, even if a graphic card is used. checkvideo: move.l pb_GfxLib(a5),a0 cmp.w #39,LIB_VERSION(a0) beq.b .known cmp.w #40,LIB_VERSION(a0) beq.b .known bra.b .nodbl .known move.l gb_copinit(a0),a0 cmp.w #$01FC,copinit_fm0(a0) ;Security check (test if really FMODE) bne.b .nodbl move.w copinit_fm0+2(a0),d0 and.w #$c000,d0 ;Mask sprite and bitplane double bit beq.b .nodbl moveq #TRUE,d0 rts .nodbl moveq #FALSE,d0 rts ;in: * d0 Frequency ;out: * d0 Closest frequency * d1 Index findfreq: lea freqlist(pc),a0 cmp.l (a0),d0 bls.b .2 .findfreq cmp.l (a0)+,d0 bhi.b .findfreq move.l -4(a0),d1 sub.l d0,d1 sub.l -8(a0),d0 cmp.l d1,d0 bhs.b .1 subq.l #4,a0 .1 subq.l #4,a0 .2 move.l (a0),d0 move.l a0,d1 sub.l #freqlist,d1 lsr.l #2,d1 rts freqlist: dc.l 4410 ; CD/10 dc.l 4800 ; DAT/10 dc.l 5513 ; CD/8 dc.l 6000 ; DAT/8 dc.l 7350 ; CD/6 dc.l 8000 ; µ- and A-Law, DAT/6 dc.l 9600 ; DAT/5 dc.l 11025 ; CD/4 dc.l 12000 ; DAT/4 dc.l 14700 ; CD/3 dc.l 16000 ; DAT/3 dc.l 17640 ; CD/2.5 dc.l 18900 dc.l 19200 ; DAT/2.5 dc.l 22050 ; CD/2 dc.l 24000 ; DAT/2 dc.l 27429 FREQUENCIES_OCS EQU (*-freqlist)/4 dc.l 29400 ; CD/1.5 dc.l 32000 ; DAT/1.5 dc.l 33075 dc.l 37800 dc.l 44100 ; CD dc.l 48000 ; DAT FREQUENCIES EQU (*-freqlist)/4 dc.l -1 * _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 ****** [driver].audio/AHIsub_FreeAudio ************************************** * * NAME * AHIsub_FreeAudio -- Deallocates the audio hardware. * * SYNOPSIS * AHIsub_FreeAudio( audioctrl ); * A2 * * void AHIsub_FreeAudio( struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * Deallocate the audio hardware and other resources allocated in * AHIsub_AllocAudio(). AHIsub_Stop() will always be called by * 'ahi.device' before this call is made. * * INPUTS * audioctrl - pointer to an AHIAudioCtrlDrv 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 .nopaula move.l d0,a3 move.l p_CalibrationTable(a3),d0 beq.b .notable move.l d0,a1 call FreeVec .notable tst.l p_audiodev(a3) bne.b .noaudiodev move.l p_audioreq(a3),a1 move.w #CMD_RESET,IO_COMMAND(a1) ;Restore audio interrupts bsr.w 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 call FreeVec .noaudioreq move.l p_audioport(a3),d0 beq.b .noaudioport move.l d0,a0 call DeleteMsgPort .noaudioport move.l pb_CardResource(a5),d0 beq .noaura move.l d0,a6 move.l p_CardHandle(a3),d0 beq .noaura move.l d0,a1 call EndCardAccess move.l p_CardHandle(a3),a1 moveq #CARDF_REMOVEHANDLE,d0 call ReleaseCard .noaura move.l pb_SysLib(a5),a6 move.l p_CardHandle(a3),d0 beq .nocardhandle move.l d0,a1 call FreeVec .nocardhandle move.l pb_MiscResource(a5),a6 tst.l p_ParPortUser(a3) bne.b .noparport moveq #MR_PARALLELPORT,d0 jsr MR_FREEMISCRESOURCE(a6) .noparport tst.l p_ParBitsUser(a3) bne.b .noparbits moveq #MR_PARALLELBITS,d0 jsr MR_FREEMISCRESOURCE(a6) .noparbits move.l pb_SysLib(a5),a6 move.l a3,a1 clr.l ahiac_DriverData(a2) call FreeVec .nopaula moveq #0,d0 popm d2-d7/a2-a6 rts ****** [driver].audio/AHIsub_Disable **************************************** * * NAME * AHIsub_Disable -- Temporary turn off audio interrupt/task * * SYNOPSIS * AHIsub_Disable( audioctrl ); * A2 * * void AHIsub_Disable( struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * 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 AHIAudioCtrlDrv 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) move.w #INTF_AUD0,CUSTOM+INTENA pop a3 rts ****** [driver].audio/AHIsub_Enable ***************************************** * * NAME * AHIsub_Enable -- Turn on audio interrupt/task * * SYNOPSIS * AHIsub_Enable( audioctrl ); * A2 * * void AHIsub_Enable( struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * 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 AHIAudioCtrlDrv 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 move.w #INTF_SETCLR!INTF_AUD0,CUSTOM+INTENA .exit pop a3 rts ****** [driver].audio/AHIsub_Start ****************************************** * * NAME * AHIsub_Start -- Starts playback or recording * * SYNOPSIS * error = AHIsub_Start( flags, audioctrl ); * D0 D0 A2 * * ULONG AHIsub_Start(ULONG, struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * 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. The buffer must * be long aligned! * B) Create/start an interrupt or task that will do 1-6 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 AHIAudioCtrlDrv *) * A1 - Set to NULL. * * 2) [Call the ahiac_PreTimer function. If it returns TRUE (Z will be * cleared so you don't have to test d0), skip step 3 and 4. This * is used to avoid overloading the CPU. This step is optional. * A2 is assumed to point to struct AHIAudioCtrlDrv. All registers * except d0 are preserved. (V4)] * * 3) Call the mixing Hook (ahiac_MixerFunc) with the following * parameters: * A0 - (struct Hook *) - The Hook itself * A2 - (struct AHIAudioCtrlDrv *) * 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 always 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 interleaved: * 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. * * For AHI V2, the type of buffer is also available in ahiac_BuffType. * It is suggested that you use this value instead. ahiac_BuffType * can be one of AHIST_M16S, AHIST_S16S, AHIST_M32S and AHIST_S32S. * * 4) 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. * * 5) [Call the ahiac_PostTimer function. A2 is assumed to point to * struct AHIAudioCtrlDrv. All registers are preserved. (V4)] * * 6) 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 with the AHISB_CANPOSTPROCESS bit set, * ahiac_BuffSize is large enough to hold two buffers. The mixing buffer * will be filled with the wet buffer first, immediately followed by the * dry buffer. I.e., ahiac_BuffSamples sample frames wet data, then * ahiac_BuffSamples sample frames dry data. The DSP fx should only be * applied to the wet buffer, and the two buffers should then be added * together. (V4) * * - 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 with neither the AHISB_MIXING nor * the AHISB_TIMING bit set, then just 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_MixerFunc, ahiac_BuffSamples, ahiac_MinBuffSamples, * ahiac_MaxBuffSamples and ahiac_BuffSize are undefined. ahiac_MixFreq * is the frequency the user wants to use for recording, if you support * that. * * * 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 - it is suggested that RecordFunc is called * at least 4 times/second for the lowers sampling rate, and more often * for higher rates), and fill it with the sampled data. The buffer must * be long aligned, and it's size must be evenly divisible by four. * 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 AHIAudioCtrlDrv *) * A1 - (struct AHIRecordMessage *) * The message should be filled as follows: * ahirm_Type - Set to AHIST_S16S. * ahirm_Buffer - A pointer to the filled buffer. * ahirm_Samples - How many sample frames stored. * You must not destroy the buffer until next time the Hook is called. * * Repeat until AHIsub_Stop() is called. * * * Note that both bits may be set when this function is called. * * INPUTS * flags - See . * audioctrl - pointer to an AHIAudioCtrlDrv 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 * The driver must be able to handle multiple calls to this routine * without preceding calls to AHIsub_Stop(). * * SEE ALSO * AHIsub_Update(), AHIsub_Stop() * ***************************************************************************** * * * AHIsub_Start: pushm d2-d7/a2-a6 move.l d0,d7 lea CUSTOM,a4 move.l ahiac_DriverData(a2),a3 btst #AHISB_PLAY,d0 beq .dont_play ** *** AHISB_PLAY ** moveq #AHISF_PLAY,d0 call AHIsub_Stop ;Stop current playback if any. call AHIsub_Update ;fill variables move.l a6,a5 move.l pb_SysLib(a5),a6 move.l ahiac_BuffSize(a2),d0 move.l #MEMF_PUBLIC|MEMF_CLEAR,d1 call AllocVec move.l d0,p_Mixbuffer(a3) beq .error_nomem move.l ahiac_MixFreq(a2),d1 bsr.w calcperiod move.w d1,p_AudPer(a3) * The init*bit? routines allocates p_DMAbuffer, sets up p_AudPtr, sets the volume, * stores the correct interrupt routines in p_PlayInt's and p_PlaySoftInt's IS_CODE. 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 .exit * Install play interrupt lea p_PlayInt(a3),a1 moveq #INTB_AUD0,d0 call SetIntVector clr.w p_DisableCount(a3) move.w #INTF_SETCLR!INTF_AUD0,INTENA(a4) ;enable move.w #INTF_SETCLR!INTF_AUD0,INTREQ(a4) ;start move.l a5,a6 .dont_play btst #AHISB_RECORD,d7 beq .dont_record ** *** AHISB_RECORD ** btst #PB_14BIT,p_Flags(a3) ;Sanity check... bne .error_unknown moveq #AHISF_RECORD,d0 call AHIsub_Stop ;Stop current recording if any. move.l a6,a5 move.l pb_SysLib(a5),a6 move.l #RECORDSAMPLES*4,d0 move.l #MEMF_PUBLIC,d1 call AllocVec move.l d0,p_RecBuffer1(a3) beq .error_nomem move.l d0,p_RecFillPtr(a3) move.w #RECORDSAMPLES,p_RecFillCount(a3) move.l #RECORDSAMPLES*4,d0 move.l #MEMF_PUBLIC,d1 call AllocVec move.l d0,p_RecBuffer2(a3) beq .error_nomem move.l ahiac_MixFreq(a2),d1 bsr.w calcperiod lsr.w #1,d1 ;Period/2 => Frequency·2 move.w d1,AUD2PER(a4) move.w d1,AUD3PER(a4) move.w p_MonitorVolume(a3),AUD2VOL(a4) move.w p_MonitorVolume(a3),AUD3VOL(a4) * Install record interrupt move.w p_Input(a3),d0 beq .parsampler cmp.w #1,d0 beq .aurasampler bra .error_unknown .parsampler tst.b p_Parallel(a3) ;Parrallel port allocated? beq .error_unknown lea p_RecIntData(a3),a1 move.l a1,IS_DATA+p_RecInt(a3) move.l #RecordInterrupt,IS_CODE+p_RecInt(a3) bra .setrecint .aurasampler tst.l p_AuraAddress(a3) ;Aura sampler allocated? beq .error_unknown lea p_RecIntDataAura(a3),a1 move.l a1,IS_DATA+p_RecInt(a3) move.l #RecordInterruptAura,IS_CODE+p_RecInt(a3) .setrecint lea p_RecInt(a3),a1 moveq #INTB_AUD3,d0 call SetIntVector move.w #DMAF_AUD2!DMAF_AUD3,DMACON(a4) ;disable DMA move.w #INTF_SETCLR!INTF_AUD3,INTENA(a4) ;enable move.w #INTF_SETCLR!INTF_AUD3,INTREQ(a4) ;start .dont_record .return moveq #AHIE_OK,d0 .exit popm d2-d7/a2-a6 rts .error_nomem moveq #AHIE_NOMEM,d0 bra.b .exit .error_unknown moveq #AHIE_UNKNOWN,d0 bra.b .exit ;in: * a2 AudioCtrl * a3 paula * a4 CUSTOM * a5 paulaBase * a6 ExecBase init8bitM: move.l #AudioInterrupt2,IS_CODE+p_PlayInt(a3) move.l #SoftInt_8bitM,IS_CODE+p_PlaySoftInt(a3) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 8 bit samples addq.l #1,d0 bclr #0,d0 ;force even 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 .nomem 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 .nomem moveq #AHIE_NOMEM,d0 rts ;in: * a2 AudioCtrl * a3 paula * a5 paulaBase * a6 ExecBase init8bitS: move.l #AudioInterrupt2,IS_CODE+p_PlayInt(a3) move.l #SoftInt_8bitS,IS_CODE+p_PlaySoftInt(a3) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 8 bit samples addq.l #1,d0 bclr #0,d0 ;force even 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 .nomem 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 .nomem moveq #AHIE_NOMEM,d0 rts ;in: * a2 AudioCtrl * a3 paula * a5 paulaBase * a6 ExecBase init14bitM: move.l #AudioInterrupt4,IS_CODE+p_PlayInt(a3) lea SoftInt_14bitM(pc),a0 tst.l p_CalibrationTable(a3) beq.b .nocalib lea SoftInt_14CbitM(pc),a0 .nocalib move.l a0,IS_CODE+p_PlaySoftInt(a3) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 16 bit samples addq.l #1,d0 bclr #0,d0 ;force even 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 .nomem 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 .nomem moveq #AHIE_NOMEM,d0 rts ;in: * a2 AudioCtrl * a3 paula * a5 paulaBase * a6 ExecBase init14bitS: move.l #AudioInterrupt4,IS_CODE+p_PlayInt(a3) lea SoftInt_14bitS(pc),a0 tst.l p_CalibrationTable(a3) beq.b .nocalib lea SoftInt_14CbitS(pc),a0 .nocalib move.l a0,IS_CODE+p_PlaySoftInt(a3) move.l ahiac_MaxBuffSamples(a2),d0 ;Max. # of 16 bit samples addq.l #1,d0 bclr #0,d0 ;force even 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 .nomem 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 .nomem moveq #AHIE_NOMEM,d0 rts ****** [driver].audio/AHIsub_Update ***************************************** * * NAME * AHIsub_Update -- Update some variables * * SYNOPSIS * AHIsub_Update( flags, audioctrl ); * D0 A2 * * void AHIsub_Update(ULONG, struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * All you have to do is to update some variables: * Mixing & timing: ahiac_PlayerFunc, ahiac_MixerFunc, ahiac_SamplerFunc, * ahiac_BuffSamples (and perhaps ahiac_PlayerFreq if you use it). * Mixing only: ahiac_PlayerFunc, ahiac_MixerFunc, ahiac_SamplerFunc and * ahiac_PlayerFreq. * Nothing: ahiac_PlayerFunc, ahiac_SamplerFunc and ahiac_PlayerFreq. * * INPUTS * flags - Currently no flags defined. * audioctrl - pointer to an AHIAudioCtrlDrv 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) move.l ahiac_PreTimer(a2),p_PreTimer(a3) move.l ahiac_PostTimer(a2),p_PostTimer(a3) call AHIsub_Enable moveq #0,d0 popm d2-d7/a2-a6 rts ****** [driver].audio/AHIsub_Stop ******************************************* * * NAME * AHIsub_Stop -- Stops playback. * * SYNOPSIS * AHIsub_Stop( flags, audioctrl ); * D0 A2 * * void AHIsub_Stop( ULONG, struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * Stop playback and/or recording, remove all resources allocated by * AHIsub_Start(). * * INPUTS * flags - See . * audioctrl - pointer to an AHIAudioCtrlDrv 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 lea CUSTOM,a4 move.l a6,a5 move.l pb_SysLib(a5),a6 move.l ahiac_DriverData(a2),a3 push d0 btst #AHISB_PLAY,d0 beq .dontplay ** *** AHISB_PLAY ** 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 #INTF_AUD0,INTENA(a4) move.w #INTF_AUD0,INTREQ(a4) ;Clear any waiting interrupts move.l #Interrupt_Dummy,IS_CODE+p_PlayInt(a3) lea p_PlayInt(a3),a1 moveq #INTB_AUD0,d0 call SetIntVector 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 .dontplay pop d0 btst #AHISB_RECORD,d0 beq .dontrecord ** *** AHISB_RECORD ** btst #PB_14BIT,p_Flags(a3) ;Sanity check... bne .dontrecord move.w #INTF_AUD3,INTENA(a4) move.w #INTF_AUD3,INTREQ(a4) ;Clear any waiting interrupts move.l #Interrupt_Dummy,IS_CODE+p_RecInt(a3) lea p_RecInt(a3),a1 moveq #INTB_AUD3,d0 call SetIntVector move.w #0,AUD2VOL(a4) move.w #0,AUD3VOL(a4) move.l p_RecBuffer1(a3),d0 beq.b .norecmem1 move.l d0,a1 clr.l p_RecBuffer1(a3) call FreeVec .norecmem1 move.l p_RecBuffer2(a3),d0 beq.b .norecmem2 move.l d0,a1 clr.l p_RecBuffer2(a3) call FreeVec .norecmem2 .dontrecord .return moveq #0,d0 popm d2-d7/a2-a6 rts ****** [driver].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.device'. * * IMPLEMENTATION * 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.device' if the driver returned AHIS_UNKNOWN. This way it is * possible to add effects that the sound card handles on its own, like * filter and echo effects. * * For what each function does, see the autodocs for 'ahi.device'. * * INPUTS * See functions in 'ahi.device'. * * NOTES * See functions in 'ahi.device'. * * SEE ALSO * ahi.device/AHI_SetEffect(), ahi.device/AHI_SetFreq(), * ahi.device/AHI_SetSound(), ahi.device/AHI_SetVol(), * ahi.device/AHI_LoadSound(), ahi.device/AHI_UnloadSound() * * ***************************************************************************** * * AHIsub_SetVol: AHIsub_SetFreq: AHIsub_SetSound: AHIsub_SetEffect: AHIsub_LoadSound: AHIsub_UnloadSound: moveq #AHIS_UNKNOWN,d0 rts ****** [driver].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 AHIAudioCtrlDrv * ); * * IMPLEMENTATION * Return the attribute based on a tag list and an AHIAudioCtrlDrv * structure, which are the same that will be passed to * AHIsub_AllocAudio() by 'ahi.device'. 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. * AHIDB_Record - Are you a sampler, too? Return TRUE or FALSE. * AHIDB_FullDuplex - Return TRUE or FALSE. * AHIDB_Realtime - Return TRUE or FALSE. * AHIDB_MaxPlaySamples - Normally, return the default. See * AHIsub_AllocAudio(), section 2. * AHIDB_MaxRecordSamples - Return the size of the buffer you fill * when recording. * * The following are associated with AHIsub_HardwareControl() and are * new for V2. * AHIDB_MinMonitorVolume * AHIDB_MaxMonitorVolume - Return the lower/upper limit for * AHIC_MonitorVolume. If unsupported but always 1.0, return * 1.0 for both. * AHIDB_MinInputGain * AHIDB_MaxInputGain - Return the lower/upper limit for * AHIC_InputGain. If unsupported but always 1.0, return 1.0 for * both. * AHIDB_MinOutputVolume * AHIDB_MaxOutputVolume - Return the lower/upper limit for * AHIC_OutputVolume. * AHIDB_Inputs - Return how many inputs you have. * AHIDB_Input - Return a short string describing the argument:th * input. Number 0 should be the default one. Example strings * can be "Line 1", "Mic", "Optical" or whatever. * AHIDB_Outputs - Return how many outputs you have. * AHIDB_Output - Return a short string describing the argument:th * output. Number 0 should be the default one. Example strings * can be "Line 1", "Headphone", "Optical" or whatever. * 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 AHIAudioCtrlDrv structure that eventually * will be fed to AHIsub_AllocAudio(), or NULL. * * NOTES * * SEE ALSO * AHIsub_AllocAudio(), AHIsub_HardwareControl(), * ahi.device/AHI_GetAudioAttrsA() * ***************************************************************************** * * AHIsub_GetAttr: pushm a3/a5-a6 move.l a6,a5 move.l pb_UtilLib(a5),a6 IFD MC020 tst.l a1 ELSE cmp.l #0,a1 ENDC beq .notaglist ;no tag list! move.l a1,a3 pushm d0-d1 move.l #AHIDB_Paula14Bit,d0 moveq #FALSE,d1 move.l a3,a0 call GetTagData tst.l d0 bne .2 * Only for AHIDB_Paula14Bit FALSE (== recording available) popm d0-d1 cmp.l #AHIDB_Record,d0 bne.b .not_record1 moveq #TRUE,d0 bra .exit .not_record1 cmp.l #AHIDB_FullDuplex,d0 bne.b .not_fullduplex1 moveq #TRUE,d0 bra .exit .not_fullduplex1 cmp.l #AHIDB_MaxRecordSamples,d0 bne.b .not_mrs1 move.l #RECORDSAMPLES,d0 bra .exit .not_mrs1 cmp.l #AHIDB_MinMonitorVolume,d0 bne.b .not_minmonvol1 moveq #0,d0 bra .exit .not_minmonvol1 cmp.l #AHIDB_MaxMonitorVolume,d0 bne.b .not_maxmonvol1 move.l #$10000,d0 bra .exit .not_maxmonvol1 cmp.l #AHIDB_MinInputGain,d0 bne.b .not_mingain1 move.l #$10000,d0 bra .exit .not_mingain1 cmp.l #AHIDB_MaxInputGain,d0 bne.b .not_maxgain1 move.l #$10000,d0 bra .exit .not_maxgain1 cmp.l #AHIDB_Inputs,d0 bne.b .not_inputs moveq #2,d0 bra .exit .not_inputs cmp.l #AHIDB_Input,d0 bne.b .not_input lsl.l #2,d1 move.l .inputs(pc,d1.w),d0 bra .exit .not_input bra .3 .inputs: dc.l .input0 dc.l .input1 .2 * Only for AHIDB_Paula14Bit TRUE (== no recording) popm d0-d1 cmp.l #AHIDB_Record,d0 bne.b .not_record2 moveq #FALSE,d0 bra .exit .not_record2 cmp.l #AHIDB_FullDuplex,d0 bne.b .not_fullduplex2 moveq #FALSE,d0 bra .exit .not_fullduplex2 .3 * Common tags (AHIDB_14bit) cmp.l #AHIDB_Bits,d0 bne.b .not_bits move.l #AHIDB_Paula14Bit,d0 moveq #FALSE,d1 move.l a3,a0 call GetTagData tst.l d0 beq.b .no14bit moveq #14,d0 skipw .no14bit moveq #8,d0 bra .exit .not_bits .notaglist cmp.l #AHIDB_Frequencies,d0 bne.b .not_freqs bsr checkvideo tst.l d0 beq.b .f move.l #FREQUENCIES,d0 bra .exit .f move.l #FREQUENCIES_OCS,d0 bra .exit .not_freqs cmp.l #AHIDB_Frequency,d0 bne.b .not_freq add.w d1,d1 add.w d1,d1 lea freqlist(pc),a0 ; move.l (a0,d1.w),d1 ; bsr calcperiod move.l (a0,d1.w),d0 bra .exit .not_freq cmp.l #AHIDB_Index,d0 bne.b .not_index move.l d1,d0 bsr findfreq move.l d1,d0 bra .exit .not_index cmp.l #AHIDB_Author,d0 bne.b .not_author lea .author(pc),a0 move.l a0,d0 bra .exit .not_author cmp.l #AHIDB_Copyright,d0 bne.b .not_copyright lea .copyright(pc),a0 move.l a0,d0 bra .exit .not_copyright cmp.l #AHIDB_Version,d0 bne.b .not_version lea IDString(pc),a0 move.l a0,d0 bra .exit .not_version cmp.l #AHIDB_Annotation,d0 bne.b .not_anno lea .anno(pc),a0 move.l a0,d0 bra .exit .not_anno cmp.l #AHIDB_Realtime,d0 bne.b .not_realtime moveq #TRUE,d0 bra .exit .not_realtime cmp.l #AHIDB_Outputs,d0 bne.b .not_outputs moveq #1,d0 bra .exit .not_outputs cmp.l #AHIDB_Output,d0 bne.b .not_output lea .output(pc),a0 move.l a0,d0 bra .exit .not_output * Unknown attribute, return default. move.l d2,d0 .exit popm a3/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 .input0 dc.b "Parallel port sampler",0 .input1 dc.b "Aura sampler",0 .output dc.b "Line",0 even ****** [driver].audio/AHIsub_HardwareControl ******************************** * * NAME * AHIsub_HardwareControl -- Modify sound card settings * * SYNOPSIS * AHIsub_HardwareControl( attribute, argument, audioctrl ); * D0 D0 D1 A2 * * LONG AHIsub_HardwareControl( ULONG, LONG, struct AHIAudioCtrlDrv * ); * * IMPLEMENTATION * Set or return the state of a particular hardware component. AHI uses * AHIsub_GetAttr() to supply the user with limits and what tags are * available. * * INPUTS * attribute - Is really a Tag and can be one of the following: * AHIC_MonitorVolume - Set the input monitor volume to argument. * AHIC_MonitorVolume_Query - Return the current input monitor * volume (argument is ignored). * * AHIC_InputGain - Set the input gain to argument. (V2) * AHIC_InputGain_Query (V2) * * AHIC_OutputVolume - Set the output volume to argument. (V2) * AHIC_OutputVolume_Query (V2) * * AHIC_Input - Use the argument:th input source (default is 0). (V2) * AHIC_Input_Query (V2) * * AHIC_Output - Use the argument:th output destination (default * is 0). (V2) * AHIC_Output_Query (V2) * * argument - What value attribute should be set to. * audioctrl - Pointer to an AHIAudioCtrlDrv structure. * * RESULT * Return the state of selected attribute. If you were asked to set * something, return TRUE. If attribute is unknown to you or unsupported, * return FALSE. * * NOTES * This call must be safe from interrupts. * * SEE ALSO * ahi.device/AHI_ControlAudioA(), AHIsub_GetAttr() * ***************************************************************************** * * AHIsub_HardwareControl: cmp.l #AHIC_MonitorVolume,d0 bne.b .dontsetmonvol move.l ahiac_DriverData(a2),a1 lsr.l #8,d1 lsr.l #2,d1 move.w d1,p_MonitorVolume(a1) bra.b .exit .dontsetmonvol cmp.l #AHIC_MonitorVolume_Query,d0 bne.b .dontgetmonvol move.l ahiac_DriverData(a2),a1 moveq #0,d0 move.w p_MonitorVolume(a1),d0 lsl.l #8,d0 lsl.l #2,d0 bra.b .quit .dontgetmonvol cmp.l #AHIC_Input,d0 bne.b .dontsetinput move.l ahiac_DriverData(a2),a1 move.w d1,p_Input(a1) bra.b .exit .dontsetinput cmp.l #AHIC_Input_Query,d0 bne.b .dontgetinput move.l ahiac_DriverData(a2),a1 moveq #0,d0 move.w p_Input(a1),d0 bra.b .quit .dontgetinput moveq #FALSE,d0 .quit rts .exit moveq #TRUE,d0 rts ;---------------------------------------------------------------------------- Interrupt_Dummy: move.w #INTF_AUD0!INTF_AUD1!INTF_AUD2!INTF_AUD3,INTREQ(a0) SoftInt_Dummy: rts ;in: * d0 scratch * d1 INTENAR && INTREQR * a0 CUSTOM * a1 &(paulaBase->p_RecIntData) * a5 &RecordInterrupt * a6 ExecBase RecordInterrupt: * This function will be executed up to 28000 times per second - that's once per * rasterline! It has to be as fast as possible. move.w #INTF_AUD2!INTF_AUD3,INTREQ(a0) ;Clear the interrupt flags moveq #0,d0 move.b _ciaa+ciaprb,d0 ;read parallel port IFD MC020 move.l convtable(pc,d0.w*4),d0 ;1 unsigned byte -> 2 signed words ELSE add.w d0,d0 add.w d0,d0 move.l convtable(pc,d0.w),d0 ENDC move.w d0,AUD2DAT(a0) ;right move.w d0,AUD3DAT(a0) ;left move.l (a1),a5 ;p_RecFillPtr move.l d0,(a5)+ ;store sample in buffer move.l a5,(a1)+ ;update pointer subq.w #1,(a1) ;p_ReqFillCount beq ri_Filled ;branch if buffer filled rts convtable CNT SET 0 REPT 256 dc.b CNT-128,CNT-128,CNT-128,CNT-128 CNT SET CNT+1 ENDR ;in: * d0 scratch * d1 INTENAR && INTREQR * a0 CUSTOM * a1 &(paulaBase->p_RecIntDataAura) * a5 &RecordInterrupt * a6 ExecBase RecordInterruptAura: move.w #INTF_AUD2!INTF_AUD3,INTREQ(a0) ;Clear the interrupt flags move.l (a1)+,a5 move.l (a5),d0 ;read aura sampler eor.l #$80008000,d0 move.l (a1),a5 ;p_RecFillPtr move.l d0,(a5)+ ;store sample in buffer move.l a5,(a1)+ ;update pointer move.w d0,d1 lsr.w #8,d0 move.b d0,d1 move.w d1,AUD2DAT(a0) ;right move.w d1,AUD3DAT(a0) ;left subq.w #1,(a1) ;p_ReqFillCount beq ri_Filled ;branch if buffer filled rts ri_Filled: * This part is only executed every RECORDSAMPLES:th time... No need to hurry. move.l 8(a1),d0 ;p_RecBuffer2-> move.l 4(a1),8(a1) ;p_RecBuffer1->p_RecBuffer2 move.l d0,4(a1) ; ->p_RecBuffer1 move.l d0,-4(a1) ;p_RecFillPtr move.w #RECORDSAMPLES,(a1) ;p_ReqFillCount move.l 12(a1),a1 ;p_RecSoftIntPtr jmp _LVOCause(a6) ;in: * d0 scratch * d1 scratch * a0 scratch * a1 paulaBase * a5 scratch RecordSoftInt: * This function is not executed many times per second and is therefore not * fully optimized... ;) push a2 move.w p_MonitorVolume(a1),CUSTOM+AUD2VOL move.w p_MonitorVolume(a1),CUSTOM+AUD3VOL move.l p_RecBuffer2(a1),p_rmBuffer(a1) move.l p_AudioCtrlP(a1),a2 lea p_RecordMessage(a1),a1 move.l ahiac_SamplerFunc(a2),a0 move.l h_Entry(a0),a5 jsr (a5) pop a2 rts ;in: * d0 scratch * d1 INTENAR && INTREQR * a0 CUSTOM * a1 &(paulaBase->p_PlayIntData) * a5 &AudioInterrupt * a6 ExecBase AudioInterrupt2: ;Two hardware channels used movem.l (a1)+,d0/d1 ;p_PlaySoftIntPtr,p_AudLen/p_AudPer move.l d1,AUD0LEN(a0) move.l d1,AUD1LEN(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 d0,a1 ;p_PlaySoftInt move.w #INTF_AUD0,INTREQ(a0) ;Clear the interrupt jmp _LVOCause(a6) ;start PlaySoftInt AudioInterrupt4: ;Four hardware channels used movem.l (a1)+,d0/d1 ;p_PlaySoftIntPtr,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_PlaySoftInt move.w #INTF_AUD0,INTREQ(a0) ;Clear the interrupt jmp _LVOCause(a6) ;start PlaySoftInt ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_PlaySoftIntData) * a5 scratch SoftInt_8bitM: pushm a2/a3/a4/a6 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3/a4/a6 jsr (a3) ;call Player Hook jsr (a4) ;call PreTimer bne .exit 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 .exit jsr (a6) ;call PostTimer, a2 ok popm a2/a3/a4/a6 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_PlaySoftIntData) * a5 scratch SoftInt_8bitS: pushm a2/a3/a4/a6 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3/a4/a6 jsr (a3) ;call Player Hook jsr (a4) ;call PreTimer bne .exit movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer movem.l (a5),a0/a3 .loop REPT 2 ;left move.b (a1),(a3)+ ;right move.b 2(a1),(a0)+ addq.l #4,a1 ENDR dbf d0,.loop .exit jsr (a6) ;call PostTimer, a2 ok popm a2/a3/a4/a6 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1,CUSTOM+DMACON rts ;in: * d0 scratch * d1 scratch * a0 scratch * a1 &(paulaBase->p_PlaySoftIntData) * a5 scratch SoftInt_14bitM: pushm a2/a3/a4/a6 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3/a4/a6 jsr (a3) ;call Player Hook jsr (a4) ;call PreTimer bne .exit movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer movem.l (a5),a0/a3/a4/a5 ;a3/a4 unused .loop REPT 2 move.b (a1)+,(a0)+ move.b (a1)+,d1 lsr.b #2,d1 move.b d1,(a5)+ ENDR dbf d0,.loop .exit jsr (a6) ;call PostTimer, a2 ok popm a2/a3/a4/a6 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_PlaySoftIntData) * a5 scratch SoftInt_14CbitM: pushm a2/a3/a4/a6 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3/a4/a6 jsr (a3) ;call Player Hook jsr (a4) ;call PreTimer bne .exit movem.l (a5),d0/a0/a1/a2/a3/a4/a5 jsr (a3) ;call Mixer Hook * convert and transfer buffer, using the table move.l (a4),a0 move.l 12(a4),a4 moveq #0,d1 .loop REPT 2 IFD MC020 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 .exit jsr (a6) ;call PostTimer, a2 ok popm a2/a3/a4/a6 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_PlaySoftIntData) * a5 scratch SoftInt_14bitS: pushm a2/a3/a4/a6 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3/a4/a6 jsr (a3) ;call Player Hook jsr (a4) ;call PreTimer bne .exit movem.l (a5),d0/a0/a1/a2/a3/a5 jsr (a3) ;call Mixer Hook push a2 * convert and transfer buffer * TODO: Use a table! movem.l (a5),a0/a2/a3/a5 .loop REPT 2 ;left move.b (a1)+,(a2)+ move.b (a1)+,d1 lsr.b #2,d1 move.b d1,(a3)+ ;right move.b (a1)+,(a0)+ move.b (a1)+,d1 lsr.b #2,d1 move.b d1,(a5)+ ENDR dbf d0,.loop pop a2 .exit jsr (a6) ;call PostTimer, a2 ok popm a2/a3/a4/a6 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_PlaySoftIntData) * a5 scratch SoftInt_14CbitS: pushm a2/a3/a4/a6 move.l a1,a5 movem.l (a5)+,a0/a1/a2/a3/a4/a6 jsr (a3) ;call Player Hook jsr (a4) ;call PreTimer bne .exit movem.l (a5),d0/a0/a1/a2/a3/a4/a5 jsr (a3) ;call Mixer Hook push a2 * convert and transfer buffer, using the table movem.l (a4),a0/a2/a3/a4 moveq #0,d1 .loop REPT 2 ;left IFD MC020 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 ;right IFD MC020 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 pop a2 .exit jsr (a6) ;call PostTimer popm a2/a3/a4/a6 move.w #DMAF_SETCLR|DMAF_AUD0|DMAF_AUD1|DMAF_AUD2|DMAF_AUD3,CUSTOM+DMACON rts EndCode: