Hri7566
/
sm64coopdx
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
sm64coopdx/include/PR/abi.h

623 lines
22 KiB
C
Raw Blame History

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* unpublished proprietary information of Silicon Graphics, Inc., and *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