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841 lines
32 KiB
C
841 lines
32 KiB
C
#ifndef _ABI_H_
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#define _ABI_H_
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/**************************************************************************
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* *
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* Copyright (C) 1994, Silicon Graphics, Inc. *
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* *
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* These coded instructions, statements, and computer programs contain *
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* unpublished proprietary information of Silicon Graphics, Inc., and *
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* are protected by Federal copyright law. They may not be disclosed *
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* to third parties or copied or duplicated in any form, in whole or *
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* in part, without the prior written consent of Silicon Graphics, Inc. *
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* *
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**************************************************************************/
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/**************************************************************************
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*
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* $Revision: 1.32 $
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* $Date: 1997/02/11 08:16:37 $
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* $Source: /exdisk2/cvs/N64OS/Master/cvsmdev2/PR/include/abi.h,v $
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*
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**************************************************************************/
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/*
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* Header file for the Audio Binary Interface.
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* This is included in the Media Binary Interface file
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* mbi.h.
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*
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* This file follows the framework used for graphics.
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*
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*/
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/* Audio commands: */
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#define A_SPNOOP 0
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#define A_ADPCM 1
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#define A_CLEARBUFF 2
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#define A_RESAMPLE 5
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#define A_SETBUFF 8
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#define A_DMEMMOVE 10
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#define A_LOADADPCM 11
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#define A_MIXER 12
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#define A_INTERLEAVE 13
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#define A_SETLOOP 15
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#ifndef VERSION_SH
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#define A_ENVMIXER 3
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#define A_LOADBUFF 4
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#define A_RESAMPLE 5
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#define A_SAVEBUFF 6
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#define A_SEGMENT 7
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#define A_SETVOL 9
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#define A_POLEF 14
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#else
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#define A_ADDMIXER 4
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#define A_RESAMPLE_ZOH 6
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#define A_INTERL 17
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#define A_ENVSETUP1 18
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#define A_ENVMIXER 19
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#define A_LOADBUFF 20
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#define A_SAVEBUFF 21
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#define A_ENVSETUP2 22
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#define A_UNK_23 23
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#define A_HILOGAIN 24
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#define A_UNK_25 25
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#define A_DUPLICATE 26
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#define A_FILTER 27
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#endif
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#define ACMD_SIZE 32
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/*
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* Audio flags
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*/
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#define A_INIT 0x01
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#define A_CONTINUE 0x00
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#define A_LOOP 0x02
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#define A_OUT 0x02
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#define A_LEFT 0x02
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#define A_RIGHT 0x00
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#define A_VOL 0x04
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#define A_RATE 0x00
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#define A_AUX 0x08
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#define A_NOAUX 0x00
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#define A_MAIN 0x00
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#define A_MIX 0x10
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/*
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* BEGIN C-specific section: (typedef's)
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*/
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#if defined(_LANGUAGE_C) || defined(_LANGUAGE_C_PLUS_PLUS)
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/*
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* Data Structures.
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*/
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int gain:16;
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unsigned int addr;
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} Aadpcm;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int gain:16;
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unsigned int addr;
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} Apolef;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int pad1:16;
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unsigned int addr;
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} Aenvelope;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:8;
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unsigned int dmem:16;
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unsigned int pad2:16;
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unsigned int count:16;
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} Aclearbuff;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:8;
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unsigned int pad2:16;
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unsigned int inL:16;
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unsigned int inR:16;
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} Ainterleave;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:24;
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unsigned int addr;
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} Aloadbuff;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int pad1:16;
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unsigned int addr;
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} Aenvmixer;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int gain:16;
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unsigned int dmemi:16;
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unsigned int dmemo:16;
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} Amixer;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int dmem2:16;
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unsigned int addr;
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} Apan;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int pitch:16;
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unsigned int addr;
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} Aresample;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int pad1:16;
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unsigned int addr;
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} Areverb;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:24;
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unsigned int addr;
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} Asavebuff;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:24;
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unsigned int pad2:2;
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unsigned int number:4;
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unsigned int base:24;
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} Asegment;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int dmemin:16;
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unsigned int dmemout:16;
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unsigned int count:16;
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} Asetbuff;
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typedef struct {
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unsigned int cmd:8;
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unsigned int flags:8;
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unsigned int vol:16;
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unsigned int voltgt:16;
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unsigned int volrate:16;
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} Asetvol;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:8;
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unsigned int dmemin:16;
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unsigned int dmemout:16;
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unsigned int count:16;
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} Admemmove;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:8;
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unsigned int count:16;
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unsigned int addr;
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} Aloadadpcm;
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typedef struct {
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unsigned int cmd:8;
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unsigned int pad1:8;
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unsigned int pad2:16;
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unsigned int addr;
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} Asetloop;
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/*
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* Generic Acmd Packet
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*/
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typedef struct {
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uintptr_t w0;
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uintptr_t w1;
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} Awords;
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typedef union {
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Awords words;
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#if IS_BIG_ENDIAN && !IS_64_BIT
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Aadpcm adpcm;
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Apolef polef;
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Aclearbuff clearbuff;
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Aenvelope envelope;
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Ainterleave interleave;
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Aloadbuff loadbuff;
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Aenvmixer envmixer;
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Aresample resample;
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Areverb reverb;
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Asavebuff savebuff;
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Asegment segment;
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Asetbuff setbuff;
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Asetvol setvol;
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Admemmove dmemmove;
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Aloadadpcm loadadpcm;
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Amixer mixer;
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Asetloop setloop;
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#endif
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long long int force_union_align; /* dummy, force alignment */
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} Acmd;
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/*
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* ADPCM State
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*/
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typedef short ADPCM_STATE[16];
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/*
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* Pole filter state
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*/
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typedef short POLEF_STATE[4];
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/*
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* Resampler state
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*/
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typedef short RESAMPLE_STATE[16];
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/*
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* Resampler constants
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*/
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#define UNITY_PITCH 0x8000
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#define MAX_RATIO 1.99996 /* within .03 cents of +1 octave */
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/*
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* Enveloper/Mixer state
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*/
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typedef short ENVMIX_STATE[40];
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/*
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* Macros to assemble the audio command list
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*/
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/*
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* Info about parameters:
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*
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* A "count" in the following macros is always measured in bytes.
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*
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* All volumes/gains are in Q1.15 signed fixed point numbers:
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* 0x8000 is the minimum volume (-100%), negating the audio curve.
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* 0x0000 is silent.
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* 0x7fff is maximum volume (99.997%).
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*
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* All DRAM addresses refer to segmented addresses. A segment table shall
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* first be set up by calling aSegment for each segment. When a DRAM
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* address is later used as parameter, the 8 high bits will be an index
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* to the segment table and the lower 24 bits are added to the base address
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* stored in the segment table for this entry. The result is the physical address.
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*
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* Transfers to/from DRAM are executed using DMA and hence follow these restrictions:
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* All DRAM addresses should be aligned by 8 bytes, or they will be
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* rounded down to the nearest multiple of 8 bytes.
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* All DRAM lengths should be aligned by 8 bytes, or they will be
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* rounded up to the nearest multiple of 8 bytes.
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*/
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/*
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* Decompresses ADPCM data.
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* Possible flags: A_INIT and A_LOOP.
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*
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* First set up internal data in DMEM:
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* aLoadADPCM(cmd++, nEntries * 16, physicalAddressOfBook)
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* aSetLoop(cmd++, physicalAddressOfLoopState) (if A_LOOP is set)
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*
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* Then before this command, call:
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* aSetBuffer(cmd++, 0, in, out, count)
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*
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* Note: count will be rounded up to the nearest multiple of 32 bytes.
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*
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* ADPCM decompression works on a block of 16 (uncompressed) samples.
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* The previous 2 samples and 9 bytes of input are decompressed to
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* 16 new samples using the code book previously loaded.
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*
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* Before the algorithm starts, the previous 16 samples are loaded according to flag:
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* A_INIT: all zeros
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* A_LOOP: the address set by aSetLoop
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* no flags: the DRAM address in the s parameter
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* These 16 samples are immediately copied to the destination address.
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*
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* The result of "count" bytes will be written after these 16 initial samples.
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* The last 16 samples written to the destination will also be written to
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* the state address in DRAM.
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*/
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#define aADPCMdec(pkt, f, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = _SHIFTL(A_ADPCM, 24, 8) | _SHIFTL(f, 16, 8); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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#define aADPCM_23(pkt, f, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = _SHIFTL(A_UNK_23, 24, 8) | _SHIFTL(f, 16, 8); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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/*
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* Not used in SM64.
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*/
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#define aPoleFilter(pkt, f, g, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = (_SHIFTL(A_POLEF, 24, 8) | _SHIFTL(f, 16, 8) | \
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_SHIFTL(g, 0, 16)); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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/*
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* Clears DMEM data, where d is address and c is count, by writing zeros.
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*
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* Note: c is rounded up to the nearest multiple of 16 bytes.
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*/
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#define aClearBuffer(pkt, d, c) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = _SHIFTL(A_CLEARBUFF, 24, 8) | _SHIFTL(d, 0, 24); \
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_a->words.w1 = (uintptr_t)(c); \
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}
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/*
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* Mixes an envelope with mono sound into 2 or 4 channels.
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* Possible flags: A_INIT, A_AUX (indicates that 4 channels should be used).
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*
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* Before this command, call:
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* aSetBuffer(cmd++, 0, inBuf, dryLeft, count)
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* aSetBuffer(cmd++, A_AUX, dryRight, wetLeft, wetRight)
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*
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* The first time (A_INIT is set), volume also needs to be set:
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* aSetVolume(cmd++, A_VOL | A_LEFT, initialVolumeLeft, 0, 0)
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* aSetVolume(cmd++, A_VOL | A_RIGHT, initialVolumeRight, 0, 0)
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* aSetVolume32(cmd++, A_RATE | A_LEFT, targetVolumeLeft, rampLeft)
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* aSetVolume32(cmd++, A_RATE | A_RIGHT, targetVolumeRight, rampRight)
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* aSetVolume(cmd++, A_AUX, dryVolume, 0, wetVolume)
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*
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* This command will now mix samples in inBuf into the destination buffers (dry and wet),
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* but with the volume increased (or decreased) from initial volumes to target volumes,
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* with the specified ramp rate. Once the target volume is reached, the volume stays
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* at that level. Before the samples are finally mixed (added) into the destination
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* buffers (dry and wet), the volume is changed according to dryVolume and wetVolume.
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*
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* Note: count will be rounded up to the nearest multiple of 16 bytes.
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* Note: the wet channels are used for reverb.
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*
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*/
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#define aEnvMixer(pkt, f, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = _SHIFTL(A_ENVMIXER, 24, 8) | _SHIFTL(f, 16, 8); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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/*
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* Interleaves two mono channels into stereo.
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*
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* First call:
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* aSetBuffer(cmd++, 0, 0, output, count)
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*
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* The count refers to the size of each input. Hence 2 * count bytes will be written out.
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* A left sample will be placed before the right sample.
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*
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* Note: count will be rounded up to the nearest multiple of 16 bytes.
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*/
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#define aInterleave(pkt, l, r) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = _SHIFTL(A_INTERLEAVE, 24, 8); \
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_a->words.w1 = _SHIFTL(l, 16, 16) | _SHIFTL(r, 0, 16); \
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}
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/*
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* Loads a buffer from DRAM to DMEM.
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*
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* First call:
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* aSetBuffer(cmd++, 0, in, 0, count)
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*
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* The in parameter to aSetBuffer is the destination in DMEM and the
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* s parameter to this command is the source in DRAM.
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*/
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#define aLoadBuffer(pkt, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = _SHIFTL(A_LOADBUFF, 24, 8); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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/*
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* Mixes audio.
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* Possible flags: no flags used, although parameter present.
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*
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* First call:
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* aSetBuffer(cmd++, 0, 0, 0, count)
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*
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* Input and output addresses are taken from the i and o parameters.
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* The volume with which the input is changed is taken from the g parameter.
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* After the volume of the input samples have been changed, the result
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* is added to the output.
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*
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* Note: count will be rounded up to the nearest multiple of 32 bytes.
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*/
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#define aMix(pkt, f, g, i, o) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = (_SHIFTL(A_MIXER, 24, 8) | _SHIFTL(f, 16, 8) | \
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_SHIFTL(g, 0, 16)); \
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_a->words.w1 = _SHIFTL(i,16, 16) | _SHIFTL(o, 0, 16); \
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}
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// Not present in the audio microcode.
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#define aPan(pkt, f, d, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = (_SHIFTL(A_PAN, 24, 8) | _SHIFTL(f, 16, 8) | \
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_SHIFTL(d, 0, 16)); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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/*
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* Resamples audio.
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* Possible flags: A_INIT, A_OUT? (not used in SM64).
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*
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* First call:
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* aSetBuffer(cmd++, 0, in, out, count)
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*
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* This command resamples the audio using the given frequency ratio (pitch)
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* using a filter that uses a window of 4 source samples. This can be used
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* either for just resampling audio to be able to be played back at a different
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* sample rate, or to change the pitch if the result is played back at
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* the same sample rate as the input.
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*
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* The frequency ratio is given in UQ1.15 fixed point format.
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* For no change in frequency, use pitch 0x8000.
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* For 1 octave up or downsampling to (roughly) half number of samples, use pitch 0xffff.
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* For 1 octave down or upsampling to double as many samples, use pitch 0x4000.
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*
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* Note: count represents the number of output sample bytes and is rounded up to
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* the nearest multiple of 16 bytes.
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*
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* The state consists of the four following source samples when the algorithm stopped as
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* well as a fractional position, and is initialized to all zeros if A_INIT is given.
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* Otherwise it is loaded from DRAM at address s.
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*
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* The algorithm starts by writing the four source samples from the state (or zero)
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* to just before the input address given. It then creates one output sample by examining
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* the four next source samples and then moving the source position zero or more
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* samples forward. The first output sample (when A_INIT is given) is always 0.
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*
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* When "count" bytes have been written, the following four source samples
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* are written to the state in DRAM as well as a fractional position.
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*/
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#define aResample(pkt, f, p, s) \
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{ \
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Acmd *_a = (Acmd *)pkt; \
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\
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_a->words.w0 = (_SHIFTL(A_RESAMPLE, 24, 8) | _SHIFTL(f, 16, 8) |\
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_SHIFTL(p, 0, 16)); \
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_a->words.w1 = (uintptr_t)(s); \
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}
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/*
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* Stores a buffer in DMEM to DRAM.
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*
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* First call:
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* aSetBuffer(cmd++, 0, 0, out, count)
|
|
*
|
|
* The out parameter to aSetBuffer is the source in DMEM and the
|
|
* s parameter to this command is the destination in DRAM.
|
|
*/
|
|
#define aSaveBuffer(pkt, s) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_SAVEBUFF, 24, 8); \
|
|
_a->words.w1 = (uintptr_t)(s); \
|
|
}
|
|
|
|
/*
|
|
* Sets up an entry in the segment table.
|
|
*
|
|
* The s parameter is a segment index, 0 to 15.
|
|
* The b parameter is the base offset.
|
|
*/
|
|
#define aSegment(pkt, s, b) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_SEGMENT, 24, 8); \
|
|
_a->words.w1 = _SHIFTL(s, 24, 8) | _SHIFTL(b, 0, 24); \
|
|
}
|
|
|
|
/*
|
|
* Sets internal DMEM buffer addresses used for later commands.
|
|
* See each command for how to use aSetBuffer.
|
|
*/
|
|
#define aSetBuffer(pkt, f, i, o, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_SETBUFF, 24, 8) | _SHIFTL(f, 16, 8) | \
|
|
_SHIFTL(i, 0, 16)); \
|
|
_a->words.w1 = _SHIFTL(o, 16, 16) | _SHIFTL(c, 0, 16); \
|
|
}
|
|
|
|
#define aInterl(pkt, f, i, o, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_INTERL, 24, 8) | _SHIFTL(f, 16, 8) | \
|
|
_SHIFTL(i, 0, 16)); \
|
|
_a->words.w1 = _SHIFTL(o, 16, 16) | _SHIFTL(c, 0, 16); \
|
|
}
|
|
|
|
/*
|
|
* Sets internal volume parameters.
|
|
* See aEnvMixer for more info.
|
|
*/
|
|
#define aSetVolume(pkt, f, v, t, r) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_SETVOL, 24, 8) | _SHIFTL(f, 16, 16) | \
|
|
_SHIFTL(v, 0, 16)); \
|
|
_a->words.w1 = _SHIFTL(t, 16, 16) | _SHIFTL(r, 0, 16); \
|
|
}
|
|
|
|
/*
|
|
* Sets the address to ADPCM loop state.
|
|
*
|
|
* The a parameter is a DRAM address.
|
|
* See aADPCMdec for more info.
|
|
*/
|
|
#define aSetLoop(pkt, a) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
_a->words.w0 = _SHIFTL(A_SETLOOP, 24, 8); \
|
|
_a->words.w1 = (uintptr_t)(a); \
|
|
}
|
|
|
|
/*
|
|
* Copies memory in DMEM.
|
|
*
|
|
* Copies c bytes from address i to address o.
|
|
*
|
|
* Note: count is rounded up to the nearest multiple of 16 bytes.
|
|
*
|
|
* Note: This acts as memcpy where 16 bytes are moved at a time, therefore
|
|
* if input and output overlap, output address should be less than input address.
|
|
*/
|
|
#define aDMEMMove(pkt, i, o, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_DMEMMOVE, 24, 8) | _SHIFTL(i, 0, 24); \
|
|
_a->words.w1 = _SHIFTL(o, 16, 16) | _SHIFTL(c, 0, 16); \
|
|
}
|
|
|
|
/*
|
|
* Loads ADPCM book from DRAM into DMEM.
|
|
*
|
|
* This command loads ADPCM table entries from DRAM to DMEM.
|
|
*
|
|
* The count parameter c should be a multiple of 16 bytes.
|
|
* The d parameter is a DRAM address.
|
|
*/
|
|
#define aLoadADPCM(pkt, c, d) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_LOADADPCM, 24, 8) | _SHIFTL(c, 0, 24); \
|
|
_a->words.w1 = (uintptr_t) (d); \
|
|
}
|
|
|
|
// This is a version of aSetVolume which takes a single 32-bit parameter
|
|
// instead of two 16-bit ones. According to AziAudio, it is used to set
|
|
// ramping values when neither bit 4 nor bit 8 is set in the flags parameter.
|
|
// It does not appear in the official abi.h header.
|
|
/*
|
|
* Sets internal volume parameters.
|
|
* See aEnvMixer for more info.
|
|
*/
|
|
#define aSetVolume32(pkt, f, v, tr) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_SETVOL, 24, 8) | _SHIFTL(f, 16, 16) | \
|
|
_SHIFTL(v, 0, 16)); \
|
|
_a->words.w1 = (uintptr_t)(tr); \
|
|
}
|
|
|
|
#ifdef VERSION_SH
|
|
#undef aLoadBuffer
|
|
#undef aSaveBuffer
|
|
#undef aMix
|
|
#undef aEnvMixer
|
|
#undef aInterleave
|
|
|
|
/*
|
|
* Mix two tracks by simple clamped addition.
|
|
*
|
|
* s: DMEM source track 1
|
|
* d: DMEM source track 2 and destination
|
|
* c: number of bytes to write (rounded down to 16 byte alignment)
|
|
*/
|
|
#define aAddMixer(pkt, s, d, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_ADDMIXER, 24, 8) | \
|
|
_SHIFTL((c) >> 4, 16, 8) | _SHIFTL(0x7fff, 0, 16)); \
|
|
_a->words.w1 = (_SHIFTL(s, 16, 16) | _SHIFTL(d, 0, 16)); \
|
|
}
|
|
|
|
/*
|
|
* Loads a buffer from DRAM to DMEM.
|
|
*
|
|
* s: DRAM source
|
|
* d: DMEM destination
|
|
* c: number of bytes to copy (rounded down to 16 byte alignment)
|
|
*/
|
|
#define aLoadBuffer(pkt, s, d, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_LOADBUFF, 24, 8) | \
|
|
_SHIFTL((c) >> 4, 16, 8) | _SHIFTL(d, 0, 16); \
|
|
_a->words.w1 = (uintptr_t)(s); \
|
|
}
|
|
|
|
/*
|
|
* Stores a buffer from DMEM to DRAM.
|
|
*
|
|
* s: DMEM source
|
|
* d: DRAM destination
|
|
* c: number of bytes to copy (rounded down to 16 byte alignment)
|
|
*/
|
|
#define aSaveBuffer(pkt, s, d, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_SAVEBUFF, 24, 8) | \
|
|
_SHIFTL((c) >> 4, 16, 8) | _SHIFTL(s, 0, 16); \
|
|
_a->words.w1 = (uintptr_t)(d); \
|
|
}
|
|
|
|
/*
|
|
* Duplicates 128 bytes of data a number of times.
|
|
*
|
|
* 128 bytes are read from source DMEM address s.
|
|
* Then c identical copies of these bytes are written to DMEM address d.
|
|
*/
|
|
#define aDuplicate(pkt, s, d, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_DUPLICATE, 24, 8) | \
|
|
_SHIFTL(c, 16, 8) | _SHIFTL(s, 0, 16)); \
|
|
_a->words.w1 = (_SHIFTL(d, 16, 16) | _SHIFTL(0x80, 0, 16)); \
|
|
}
|
|
|
|
/*
|
|
* Fast resample.
|
|
*
|
|
* Before this command, call:
|
|
* aSetBuffer(cmd++, 0, in, out, count)
|
|
*
|
|
* This works like the other resample command but just takes the "nearest" sample,
|
|
* instead of a function of the four nearest samples.
|
|
*/
|
|
#define aResampleZoh(pkt, pitch, start_fract) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_RESAMPLE_ZOH, 24, 8) | \
|
|
_SHIFTL(pitch, 0, 16)); \
|
|
_a->words.w1 = _SHIFTL(start_fract, 0, 16); \
|
|
}
|
|
|
|
/*
|
|
* Mixes audio.
|
|
*
|
|
* Input and output addresses are taken from the i and o parameters.
|
|
* The volume with which the input is changed is taken from the g parameter.
|
|
* After the volume of the input samples have been changed, the result
|
|
* is added to the output.
|
|
*
|
|
* Note: count is first rounded down to the nearest multiple of 16 bytes
|
|
* and then rounded up to the nearest multiple of 32 bytes.
|
|
*/
|
|
#define aMix(pkt, g, i, o, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_MIXER, 24, 8) | \
|
|
_SHIFTL((c) >> 4, 16, 8) | _SHIFTL(g, 0, 16)); \
|
|
_a->words.w1 = _SHIFTL(i, 16, 16) | _SHIFTL(o, 0, 16); \
|
|
}
|
|
|
|
#define aEnvSetup1(pkt, a, b, c, d) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_ENVSETUP1, 24, 8) | \
|
|
_SHIFTL(a, 16, 8) | _SHIFTL(b, 0, 16)); \
|
|
_a->words.w1 = _SHIFTL(c, 16, 16) | _SHIFTL(d, 0, 16); \
|
|
}
|
|
|
|
#define aEnvSetup2(pkt, volLeft, volRight) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_ENVSETUP2, 24, 8); \
|
|
_a->words.w1 = _SHIFTL(volLeft, 16, 16) | \
|
|
_SHIFTL(volRight, 0, 16); \
|
|
}
|
|
|
|
#define aEnvMixer(pkt, inBuf, nSamples, bit1, bit2, bit3, dryLeft, dryRight, wetLeft, wetRight) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_ENVMIXER, 24, 8) | \
|
|
_SHIFTL((inBuf) >> 4, 16, 8) | \
|
|
_SHIFTL(nSamples, 8, 8)) | \
|
|
_SHIFTL(bit1, 2, 1) | _SHIFTL(bit2, 1, 1) | \
|
|
_SHIFTL(bit3, 0, 1); \
|
|
_a->words.w1 = _SHIFTL((dryLeft) >> 4, 24, 8) | \
|
|
_SHIFTL((dryRight) >> 4, 16, 8) | \
|
|
_SHIFTL((wetLeft) >> 4, 8, 8) | \
|
|
_SHIFTL((wetRight) >> 4, 0, 8); \
|
|
}
|
|
|
|
#define aInterleave(pkt, o, l, r, c) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_INTERLEAVE, 24, 8) | \
|
|
_SHIFTL((c) >> 4, 16, 8) | _SHIFTL(o, 0, 16); \
|
|
_a->words.w1 = _SHIFTL(l, 16, 16) | _SHIFTL(r, 0, 16); \
|
|
}
|
|
|
|
// countOrBuf meaning depends on flag
|
|
#define aFilter(pkt, f, countOrBuf, addr) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_FILTER, 24, 8) | _SHIFTL((f), 16, 8) | \
|
|
_SHIFTL((countOrBuf), 0, 16); \
|
|
_a->words.w1 = (uintptr_t)(addr); \
|
|
}
|
|
|
|
#define aHilogain(pkt, id, buflen, i) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = _SHIFTL(A_HILOGAIN, 24, 8) | \
|
|
_SHIFTL((id), 16, 8) | _SHIFTL((buflen), 0, 16); \
|
|
_a->words.w1 = _SHIFTL((i), 16, 16); \
|
|
}
|
|
|
|
#define aUnknown25(pkt, f, g, i, o) \
|
|
{ \
|
|
Acmd *_a = (Acmd *)pkt; \
|
|
\
|
|
_a->words.w0 = (_SHIFTL(A_UNK_25, 24, 8) | \
|
|
_SHIFTL((f), 16, 8) | _SHIFTL((g), 0, 16)); \
|
|
_a->words.w1 = _SHIFTL((i), 16, 16) | _SHIFTL((o), 0, 16); \
|
|
}
|
|
|
|
#endif
|
|
|
|
#endif /* _LANGUAGE_C */
|
|
|
|
#endif /* !_ABI_H_ */
|