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goldeneye_src/src/audi.c
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Larry Ficken 858d640da7 update public from private
2021-03-08 12:58:16 -06:00

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#include "ultra64.h"
#include "sched.h"
#include "audi.h"
/**
* @file audi.c
* This file contains audio code. Starts main audio thread, handles some audio DMA.
*/
// 0x5622 = 22050
#define OUTPUT_RATE 0x5622
#define MAYBE_FRAME_RATE 60
#define FRAMES_PER_FIELD_AS_POW2 1
#define AUDIO_FRAME_MESSAGE_QUEUE_SIZE 8
#define AUDIO_REPLY_MESSAGE_QUEUE_SIZE 8
#define AUDIO_DMA_IO_QUEUE_SIZE 64
#define AUDIO_DMA_QUEUE_SIZE 66
#define AUDIO_DMA_MAX_BUFFER_LENGTH 0x200
#define NUMBER_OUTPUT_BUFFERS 3
#define NUMBER_ACMD_LISTS 2
#define MAX_ACMD_SIZE 3000
#define NUMBER_DMA_BUFFERS 64
#define EXTRA_SAMPLES 0x25
#define AUDIO_FRAME_MESSAGE_QUEUE_SIZE 8
#define AUDIO_REPLY_MESSAGE_QUEUE_SIZE 8
#define MAIN_QUIT_MESSAGE 10
#define AUDIO_MANAGER_COUNT_INTERVAL 0xf0
extern long long int rspbootTextStart[];
extern long long int gsp3DTextStart[];
extern long long int aspMainTextStart[];
extern long long int aspMainDataStart[];
extern u8 sp_audi[];
/**
* Copied from the n64devkit audio examples.
* sizeof(struct DMABuffer_s) == 0x14 (20)
*/
typedef struct DMABuffer_s {
/**
* 0x0.
*/
ALLink node;
/**
* 0x8.
*/
int startAddr;
/**
* 0xc.
*/
u32 lastFrame;
/**
* 0x10.
*/
u8* ptr;
} DMABuffer;
/**
* Copied from the n64devkit audio examples.
* sizeof(struct DMAState_s) == 0xc (12).
*/
typedef struct DMAState_s {
/**
* This was defined (in the devkit) as u8 (and code expects a byte), but the size
* of the struct and offset for firstUsed seems to make this u32/s32.
* I'm adding the union to make this explicit.
* 0x0.
*/
union {
u8 initialized;
s32 _unusedAlign;
} u;
/**
* 0x4.
*/
DMABuffer *firstUsed;
/**
* 0x8.
*/
DMABuffer *firstFree;
} DMAState;
/**
* Copied from the n64devkit audio examples.
*/
typedef union AudioMessage_u {
struct {
s16 type;
} gen;
struct {
s16 type;
struct AudioInfo_s *info;
} done;
OSScMsg app;
} AudioMessage;
/**
* Modified from n64devkit example.
* sizeof(struct _DMAState) == 0xc (12).
*/
typedef struct AudioInfo_s {
/**
* Output data pointer.
* 0x0.
*/
s16 *data;
/**
* # of samples synthesized in this frame
* 0x4.
*/
s16 frameSamples;
/**
* scheduler structure
* 0x8
*/
OSScTask task;
} AudioInfo;
// unknown purpose
u32 D_800230F0 = 0;
u32 g_AudioFrameCount = 0;
u32 g_NextDMa = 0;
u32 g_CurrentAcmdList = 0;
/*
* This macro is used/defined in both libultra and libnaudio
*/
#define ms *(((s32)((f32)44.1)) & ~0x7)
#define CUSTOM_FX_SECTION_COUNT 6
#define CUSTOM_FX_SECTION_SIZE 8
/*
* Following the libultra and libnaudio naming convention ...
*/
s32 CUSTOM_FX_PARAMS_N[CUSTOM_FX_SECTION_COUNT * CUSTOM_FX_SECTION_SIZE + 2] = {
/* sections length */
6, 160 ms,
/* chorus chorus filter
input output fbcoef ffcoef gain rate depth coef */
0, 4 ms, 9830, -9830, 0, 0, 0, 0,
4 ms, 8 ms, 9830, -9830, 0x2B84, 0, 0, 0x2500,
20 ms, 64 ms, 16384, -16384, 0x11EB, 0, 0, 0x3000,
80 ms, 140 ms, 16384, -16384, 0x11EB, 0, 0, 0x3500,
84 ms, 120 ms, 8192, -8192, 0, 0, 0, 0x4000,
0, 148 ms, 13000, -13000, 0, 0x017C, 0xA, 0x4500
};
s32 g_FirstTime = 1;
/*bss needs fixing */
s32 dword_CODE_bss_8005E4B0[2];
/**
* Address 8005E4B8.
* (type is u64)
* Used in amMain.
* This looks like it stores the largest sDeltaTime between
* counts of AUDIO_MANAGER_COUNT_INTERVAL.
*/
OSTime g_LargestDeltaTime;
/**
* Address 8005E4C0.
* (type is u64)
* Used in amMain.
* Stores the elpased time of main loop (difference between sEndTime and sStartTime).
*/
#ifdef NONMATCHING
OSTime g_DeltaTime;
#else
s32 dword_CODE_bss_8005E4C0;
s32 dword_CODE_bss_8005E4C4;
#endif
/**
* Address 8005E4C8.
* Every AUDIO_MANAGER_COUNT_INTERVAL number of events, the average for sDeltaTimeSum
* is computed and stored here.
*/
#ifdef NONMATCHING
u64 g_DeltaAverage;
#else
s32 dword_CODE_bss_8005E4C8;
s32 dword_CODE_bss_8005E4CC;
#endif
/**
* Address 8005E4D0.
* Tracks the sum total elapsed time. Reset every AUDIO_MANAGER_COUNT_INTERVAL.
*/
u64 g_DeltaTimeSum;
/**
* Address 8005E4D8.
* (type is u64)
* Used in amMain.
* Stores the time at the start of the loop.
*/
OSTime g_StartTime;
/**
* Address 8005E4E0.
* (type is u64)
* Used in amMain.
* Stores the time after primary processing is done.
*/
OSTime g_EndTime;
/**
* Unknown / unused
*/
char dword_CODE_bss_8005E4E8[0x30];
/**
* Address 8005e518.
* sizeof(struct AudioManager_s) == 0x288 (648)
*/
struct AudioManager_s {
/**
* 0.
*/
Acmd *cmdList[NUMBER_ACMD_LISTS];
/**
* 0x8.
*/
AudioInfo *audioInfo[NUMBER_OUTPUT_BUFFERS];
/**
* 0x14.
*/
u32 numberOutputBuffers;
/**
* 0x18.
*/
OSThread audioThread;
/**
* 0x1c8.
*/
OSMesgQueue frameMessageQueue;
/**
* 0x1e0.
*/
OSMesg frameMessageBuffer[AUDIO_FRAME_MESSAGE_QUEUE_SIZE];
/**
* 0x200.
*/
OSMesgQueue replyMessageQueue;
/**
* 0x218.
*/
OSMesg replyMessageBuffer[AUDIO_REPLY_MESSAGE_QUEUE_SIZE];
/**
* 0x238
*/
ALGlobals g;
} g_AudioManager;
/**
* Address 0x8005e7a0.
*/
OSScClient g_AudioClient[2];
/**
* Address 0x8005e7b0.
*/
DMAState g_DmaState;
DMABuffer g_DmaBuffers[NUMBER_DMA_BUFFERS];
u32 g_MinFrameSize;
u32 g_FrameSize;
u32 g_MaxFrameSize;
s32 g_CommandLength;
OSIoMesg g_DmaIOMessageBuffer[AUDIO_DMA_IO_QUEUE_SIZE];
OSMesgQueue g_DmaMessageQueue;
OSMesg g_DmaMessageBuffer[AUDIO_DMA_QUEUE_SIZE];
// Forward declarations
s32 amDmaCallback(s32 addr, s32 len, void* state);
void amClearDmaBuffers(void);
void amHandleFrameMessage(AudioInfo *info, AudioInfo *lastInfo);
void amHandleDoneMessage(AudioInfo *info);
void amMain(void* arg);
ALDMAproc amDmaNew(DMAState** state);
/**
* Address 29D0 70001BD0
*
* Looks to be loosely based on method
* amCreateAudioMgr
* from the n64devkit.
*
* @param alconf hw setup/config.
*/
void amCreateAudioManager(ALSynConfig* alconf)
{
u32 j;
f32 fsize;
alconf->dmaproc = &amDmaNew;
alconf->outputRate = osAiSetFrequency(OUTPUT_RATE);
fsize = (f32) ((alconf->outputRate << FRAMES_PER_FIELD_AS_POW2) / (f32)MAYBE_FRAME_RATE);
g_FrameSize = (u32) fsize;
if (g_FrameSize < fsize)
{
g_FrameSize++;
}
// This rounds up to the next multiple of 16.
if (g_FrameSize & 0xf)
{
g_FrameSize = (g_FrameSize & ~0xf) + 0x10;
}
g_MinFrameSize = (u32)(g_FrameSize - 0x10);
g_MaxFrameSize = (u32)(g_FrameSize + EXTRA_SAMPLES + 0x10);
if (alconf->fxType == AL_FX_CUSTOM)
{
s32 sp48[CUSTOM_FX_SECTION_COUNT * CUSTOM_FX_SECTION_SIZE + 2] = CUSTOM_FX_PARAMS_N;
alconf->params = sp48;
alInit(&g_AudioManager.g, alconf);
}
else
{
alInit(&g_AudioManager.g, alconf);
}
for (j=0; j < NUMBER_OUTPUT_BUFFERS; j++)
{
g_AudioManager.audioInfo[j] = (AudioInfo *)alHeapDBAlloc(0, 0, alconf->heap, 1, sizeof(AudioInfo));
g_AudioManager.audioInfo[j]->data = (s16*)alHeapDBAlloc(0, 0, alconf->heap, 1, g_MaxFrameSize * 4);
}
osCreateMesgQueue(&g_AudioManager.replyMessageQueue, (OSMesg*)&g_AudioManager.replyMessageBuffer, AUDIO_REPLY_MESSAGE_QUEUE_SIZE);
osCreateMesgQueue(&g_AudioManager.frameMessageQueue, (OSMesg*)&g_AudioManager.frameMessageBuffer, AUDIO_FRAME_MESSAGE_QUEUE_SIZE);
osCreateMesgQueue(&g_DmaMessageQueue, (OSMesg*)&g_DmaMessageBuffer, AUDIO_DMA_IO_QUEUE_SIZE);
g_DmaBuffers[0].node.prev = NULL;
g_DmaBuffers[0].node.next = NULL;
for (j=0; (s32)j < NUMBER_DMA_BUFFERS - 1; j++)
{
alLink((ALLink*)&g_DmaBuffers[j+1], (ALLink*)&g_DmaBuffers[j]);
g_DmaBuffers[j].ptr = (void*)alHeapDBAlloc(0, 0, alconf->heap, 1, AUDIO_DMA_MAX_BUFFER_LENGTH);
}
// last buffer already linked, but still needs buffer
g_DmaBuffers[j].ptr = (void*)alHeapDBAlloc(0, 0, alconf->heap, 1, AUDIO_DMA_MAX_BUFFER_LENGTH);
for (j=0; j < NUMBER_ACMD_LISTS; j++)
{
g_AudioManager.cmdList[j] = (Acmd *)alHeapDBAlloc(0, 0, alconf->heap, 1, MAX_ACMD_SIZE * sizeof(Acmd));
}
osCreateThread(&g_AudioManager.audioThread, 4, &amMain, 0, (void*)set_stack_entry((u8*)(&sp_audi), 0x1000), 0x14);
}
/**
* 2B58 70001F58
* insert sound manager thread
* redirect to 7000D580: A0=8005E530
*/
void amStartAudioThread(void)
{
osStartThread(&g_AudioManager.audioThread);
}
#ifdef NONMATCHING
/**
* 2B7C 70001F7C
* Looks to be loosely based on method
* __amMain
* from the n64devkit. This method makes some kind of video calls,
* but also does some kind of debug tracking of the time spent between
* beginning and end of processing.
*
* @param arg unused.
*
* decomp status:
* - compiles: yes
* - stack resize: ok
* - identical instructions: fail
* - identical registers: fail
*
* notes: It looks like there are a few issues.
* 1) The static variables are being loaded into the wrong registers.
* This is sEndTime, sStartTime, sLargestDeltaTime.
* This causes trickle down differences.
* 2) Inside the first if block, registers are saved onto the stack,
* but the stack locations differ.
* This causes trickle down differences.
* 3) localDelta is probably a red herring, it makes the code close, but ...
* 4) g_DeltaAverage and g_DeltaTime don't seem to be written correctly as u64.
* Compare:
4e8: 3c010000 lui at,0x0 4e8: R_MIPS_HI16 dword_CODE_bss_8005E4C0
4ec: 006bc823 subu t9,v1,t3
4f0: ac390000 sw t9,0(at) 4f0: R_MIPS_LO16 dword_CODE_bss_8005E4C4
4f4: ac380000 sw t8,0(at) 4f4: R_MIPS_LO16 dword_CODE_bss_8005E4C0
to
4e8: 3c010000 lui at,0x0 4e8: R_MIPS_HI16 g_DeltaTime
4ec: 006bc823 subu t9,v1,t3
4f0: ac390004 sw t9,4(at) 4f0: R_MIPS_LO16 g_DeltaTime
4f4: ac380000 sw t8,0(at) 4f4: R_MIPS_LO16 g_DeltaTime
*/
void amMain(void* arg)
{
s32 counter = 0;
s32 done = 0;
AudioMessage *msg = 0;
AudioInfo *lastInfo = 0;
u64 localDelta;
osScAddClient(&sc, &(g_AudioClient[0]), &(g_AudioManager.frameMessageQueue), (void*)1);
while (!done)
{
osRecvMesg(&g_AudioManager.frameMessageQueue, (OSMesg *)&msg, OS_MESG_BLOCK);
switch (msg->gen.type)
{
case OS_SC_RETRACE_MSG:
{
g_StartTime = osGetTime();
speedGraphVideoRelated_3(0x30000);
amHandleFrameMessage(g_AudioManager.audioInfo[g_AudioFrameCount % 3U], lastInfo);
counter++;
speedGraphVideoRelated_3(0x60000);
g_EndTime = osGetTime();
localDelta = g_EndTime - g_StartTime;
g_DeltaTime = localDelta;
if ((counter % AUDIO_MANAGER_COUNT_INTERVAL) == 0)
{
g_DeltaAverage = g_DeltaTimeSum / AUDIO_MANAGER_COUNT_INTERVAL;
localDelta = g_EndTime - g_StartTime;
g_DeltaTimeSum = 0U;
g_LargestDeltaTime = 0U;
}
else
{
g_DeltaTimeSum = g_DeltaTimeSum + g_EndTime - g_StartTime;
}
if (g_LargestDeltaTime < localDelta)
{
g_LargestDeltaTime = localDelta;
}
osRecvMesg(&g_AudioManager.replyMessageQueue, (OSMesg *)&lastInfo, OS_MESG_BLOCK);
amHandleDoneMessage(lastInfo);
}
break;
case 5:
done = 1;
break;
case MAIN_QUIT_MESSAGE:
done = 1;
break;
default:
break;
}
}
alClose(&(g_AudioManager.g));
}
#else
GLOBAL_ASM(
.text
glabel amMain
/* 002B7C 70001F7C 27BDFF90 */ addiu $sp, $sp, -0x70
/* 002B80 70001F80 AFB60030 */ sw $s6, 0x30($sp)
/* 002B84 70001F84 3C168006 */ lui $s6, %hi(g_AudioManager+0x1C8)
/* 002B88 70001F88 AFA40070 */ sw $a0, 0x70($sp)
/* 002B8C 70001F8C 26D6E6E0 */ addiu $s6, %lo(g_AudioManager+0x1C8) # addiu $s6, $s6, -0x1920
/* 002B90 70001F90 AFBF003C */ sw $ra, 0x3c($sp)
/* 002B94 70001F94 AFB20020 */ sw $s2, 0x20($sp)
/* 002B98 70001F98 AFB1001C */ sw $s1, 0x1c($sp)
/* 002B9C 70001F9C 3C048006 */ lui $a0, %hi(sc)
/* 002BA0 70001FA0 3C058006 */ lui $a1, %hi(g_AudioClient)
/* 002BA4 70001FA4 AFBE0038 */ sw $fp, 0x38($sp)
/* 002BA8 70001FA8 AFB70034 */ sw $s7, 0x34($sp)
/* 002BAC 70001FAC AFB5002C */ sw $s5, 0x2c($sp)
/* 002BB0 70001FB0 AFB40028 */ sw $s4, 0x28($sp)
/* 002BB4 70001FB4 AFB30024 */ sw $s3, 0x24($sp)
/* 002BB8 70001FB8 AFB00018 */ sw $s0, 0x18($sp)
/* 002BBC 70001FBC 00008825 */ move $s1, $zero
/* 002BC0 70001FC0 00009025 */ move $s2, $zero
/* 002BC4 70001FC4 AFA00064 */ sw $zero, 0x64($sp)
/* 002BC8 70001FC8 AFA00060 */ sw $zero, 0x60($sp)
/* 002BCC 70001FCC 24A5E7A0 */ addiu $a1, %lo(g_AudioClient) # addiu $a1, $a1, -0x1860
/* 002BD0 70001FD0 2484DA40 */ addiu $a0, %lo(sc) # addiu $a0, $a0, -0x25c0
/* 002BD4 70001FD4 02C03025 */ move $a2, $s6
/* 002BD8 70001FD8 0C000305 */ jal osScAddClient
/* 002BDC 70001FDC 24070001 */ li $a3, 1
/* 002BE0 70001FE0 3C158006 */ lui $s5, %hi(g_EndTime)
/* 002BE4 70001FE4 3C148006 */ lui $s4, %hi(g_StartTime)
/* 002BE8 70001FE8 3C138006 */ lui $s3, %hi(g_LargestDeltaTime)
/* 002BEC 70001FEC 3C108006 */ lui $s0, %hi(g_DeltaTimeSum)
/* 002BF0 70001FF0 2610E4D0 */ addiu $s0, %lo(g_DeltaTimeSum) # addiu $s0, $s0, -0x1b30
/* 002BF4 70001FF4 2673E4B8 */ addiu $s3, %lo(g_LargestDeltaTime) # addiu $s3, $s3, -0x1b48
/* 002BF8 70001FF8 2694E4D8 */ addiu $s4, %lo(g_StartTime) # addiu $s4, $s4, -0x1b28
/* 002BFC 70001FFC 26B5E4E0 */ addiu $s5, %lo(g_EndTime) # addiu $s5, $s5, -0x1b20
/* 002C00 70002000 241E0001 */ li $fp, 1
/* 002C04 70002004 27B70064 */ addiu $s7, $sp, 0x64
/* 002C08 70002008 02C02025 */ move $a0, $s6
.L7000200C:
/* 002C0C 7000200C 02E02825 */ move $a1, $s7
/* 002C10 70002010 0C003774 */ jal osRecvMesg
/* 002C14 70002014 03C03025 */ move $a2, $fp
/* 002C18 70002018 8FAE0064 */ lw $t6, 0x64($sp)
/* 002C1C 7000201C 24010005 */ li $at, 5
/* 002C20 70002020 85C40000 */ lh $a0, ($t6)
/* 002C24 70002024 109E0007 */ beq $a0, $fp, .L70002044
/* 002C28 70002028 00000000 */ nop
/* 002C2C 7000202C 10810071 */ beq $a0, $at, .L700021F4
/* 002C30 70002030 2401000A */ li $at, 10
/* 002C34 70002034 50810072 */ beql $a0, $at, .L70002200
/* 002C38 70002038 03C09025 */ move $s2, $fp
/* 002C3C 7000203C 10000070 */ b .L70002200
/* 002C40 70002040 00000000 */ nop
.L70002044:
/* 002C44 70002044 0C003AEC */ jal osGetTime
/* 002C48 70002048 00000000 */ nop
/* 002C4C 7000204C AE820000 */ sw $v0, ($s4)
/* 002C50 70002050 AE830004 */ sw $v1, 4($s4)
/* 002C54 70002054 0C000A15 */ jal speedGraphVideoRelated_3
/* 002C58 70002058 3C040003 */ lui $a0, 3
/* 002C5C 7000205C 3C0F8002 */ lui $t7, %hi(g_AudioFrameCount)
/* 002C60 70002060 8DEF30F4 */ lw $t7, %lo(g_AudioFrameCount)($t7)
/* 002C64 70002064 24010003 */ li $at, 3
/* 002C68 70002068 3C048006 */ lui $a0, %hi(g_AudioManager+8)
/* 002C6C 7000206C 01E1001B */ divu $zero, $t7, $at
/* 002C70 70002070 0000C010 */ mfhi $t8
/* 002C74 70002074 0018C880 */ sll $t9, $t8, 2
/* 002C78 70002078 00992021 */ addu $a0, $a0, $t9
/* 002C7C 7000207C 8C84E520 */ lw $a0, %lo(g_AudioManager+8)($a0)
/* 002C80 70002080 0C000891 */ jal amHandleFrameMessage
/* 002C84 70002084 8FA50060 */ lw $a1, 0x60($sp)
/* 002C88 70002088 26310001 */ addiu $s1, $s1, 1
/* 002C8C 7000208C 0C000A15 */ jal speedGraphVideoRelated_3
/* 002C90 70002090 3C040006 */ lui $a0, 6
/* 002C94 70002094 0C003AEC */ jal osGetTime
/* 002C98 70002098 00000000 */ nop
/* 002C9C 7000209C 8E8A0000 */ lw $t2, ($s4)
/* 002CA0 700020A0 8E8B0004 */ lw $t3, 4($s4)
/* 002CA4 700020A4 AEA20000 */ sw $v0, ($s5)
/* 002CA8 700020A8 004AC023 */ subu $t8, $v0, $t2
/* 002CAC 700020AC 006B082B */ sltu $at, $v1, $t3
/* 002CB0 700020B0 0301C023 */ subu $t8, $t8, $at
/* 002CB4 700020B4 AEA30004 */ sw $v1, 4($s5)
/* 002CB8 700020B8 3C018006 */ lui $at, %hi(dword_CODE_bss_8005E4C0)
/* 002CBC 700020BC 006BC823 */ subu $t9, $v1, $t3
/* 002CC0 700020C0 AC39E4C4 */ sw $t9, %lo(dword_CODE_bss_8005E4C4)($at)
/* 002CC4 700020C4 AC38E4C0 */ sw $t8, %lo(dword_CODE_bss_8005E4C0)($at)
/* 002CC8 700020C8 240100F0 */ li $at, 240
/* 002CCC 700020CC 0221001A */ div $zero, $s1, $at
/* 002CD0 700020D0 00004010 */ mfhi $t0
/* 002CD4 700020D4 AFB90044 */ sw $t9, 0x44($sp)
/* 002CD8 700020D8 AFB80040 */ sw $t8, 0x40($sp)
/* 002CDC 700020DC AFA20050 */ sw $v0, 0x50($sp)
/* 002CE0 700020E0 AFA30054 */ sw $v1, 0x54($sp)
/* 002CE4 700020E4 AFAA0048 */ sw $t2, 0x48($sp)
/* 002CE8 700020E8 1500001C */ bnez $t0, .L7000215C
/* 002CEC 700020EC AFAB004C */ sw $t3, 0x4c($sp)
/* 002CF0 700020F0 8E040000 */ lw $a0, ($s0)
/* 002CF4 700020F4 8E050004 */ lw $a1, 4($s0)
/* 002CF8 700020F8 24060000 */ li $a2, 0
/* 002CFC 700020FC 0C003B2A */ jal __ull_div
/* 002D00 70002100 240700F0 */ li $a3, 240
/* 002D04 70002104 3C018006 */ lui $at, %hi(dword_CODE_bss_8005E4CC)
/* 002D08 70002108 AC22E4C8 */ sw $v0, %lo(dword_CODE_bss_8005E4C8)($at)
/* 002D0C 7000210C AC23E4CC */ sw $v1, %lo(dword_CODE_bss_8005E4CC)($at)
/* 002D10 70002110 8E990004 */ lw $t9, 4($s4)
/* 002D14 70002114 8E980000 */ lw $t8, ($s4)
/* 002D18 70002118 8EAF0004 */ lw $t7, 4($s5)
/* 002D1C 7000211C 8EAE0000 */ lw $t6, ($s5)
/* 002D20 70002120 240A0000 */ li $t2, 0
/* 002D24 70002124 01F9082B */ sltu $at, $t7, $t9
/* 002D28 70002128 01D84023 */ subu $t0, $t6, $t8
/* 002D2C 7000212C 01014023 */ subu $t0, $t0, $at
/* 002D30 70002130 240B0000 */ li $t3, 0
/* 002D34 70002134 240C0000 */ li $t4, 0
/* 002D38 70002138 240D0000 */ li $t5, 0
/* 002D3C 7000213C 01F94823 */ subu $t1, $t7, $t9
/* 002D40 70002140 AFA90044 */ sw $t1, 0x44($sp)
/* 002D44 70002144 AE6D0004 */ sw $t5, 4($s3)
/* 002D48 70002148 AE6C0000 */ sw $t4, ($s3)
/* 002D4C 7000214C AE0B0004 */ sw $t3, 4($s0)
/* 002D50 70002150 AFA80040 */ sw $t0, 0x40($sp)
/* 002D54 70002154 10000011 */ b .L7000219C
/* 002D58 70002158 AE0A0000 */ sw $t2, ($s0)
.L7000215C:
/* 002D5C 7000215C 8E0B0004 */ lw $t3, 4($s0)
/* 002D60 70002160 8FAD0054 */ lw $t5, 0x54($sp)
/* 002D64 70002164 8E0A0000 */ lw $t2, ($s0)
/* 002D68 70002168 8FAC0050 */ lw $t4, 0x50($sp)
/* 002D6C 7000216C 016D7821 */ addu $t7, $t3, $t5
/* 002D70 70002170 8FB80048 */ lw $t8, 0x48($sp)
/* 002D74 70002174 8FB9004C */ lw $t9, 0x4c($sp)
/* 002D78 70002178 01ED082B */ sltu $at, $t7, $t5
/* 002D7C 7000217C 002A7021 */ addu $t6, $at, $t2
/* 002D80 70002180 01CC7021 */ addu $t6, $t6, $t4
/* 002D84 70002184 01D84023 */ subu $t0, $t6, $t8
/* 002D88 70002188 01F9082B */ sltu $at, $t7, $t9
/* 002D8C 7000218C 01014023 */ subu $t0, $t0, $at
/* 002D90 70002190 01F94823 */ subu $t1, $t7, $t9
/* 002D94 70002194 AE090004 */ sw $t1, 4($s0)
/* 002D98 70002198 AE080000 */ sw $t0, ($s0)
.L7000219C:
/* 002D9C 7000219C 8E6A0000 */ lw $t2, ($s3)
/* 002DA0 700021A0 8FAC0040 */ lw $t4, 0x40($sp)
/* 002DA4 700021A4 8E6B0004 */ lw $t3, 4($s3)
/* 002DA8 700021A8 8FAD0044 */ lw $t5, 0x44($sp)
/* 002DAC 700021AC 018A082B */ sltu $at, $t4, $t2
/* 002DB0 700021B0 14200008 */ bnez $at, .L700021D4
/* 002DB4 700021B4 3C048006 */ lui $a0, %hi(g_AudioManager+0x200)
/* 002DB8 700021B8 014C082B */ sltu $at, $t2, $t4
/* 002DBC 700021BC 14200003 */ bnez $at, .L700021CC
/* 002DC0 700021C0 016D082B */ sltu $at, $t3, $t5
/* 002DC4 700021C4 10200003 */ beqz $at, .L700021D4
/* 002DC8 700021C8 00000000 */ nop
.L700021CC:
/* 002DCC 700021CC AE6C0000 */ sw $t4, ($s3)
/* 002DD0 700021D0 AE6D0004 */ sw $t5, 4($s3)
.L700021D4:
/* 002DD4 700021D4 2484E718 */ addiu $a0, %lo(g_AudioManager+0x200) # addiu $a0, $a0, -0x18e8
/* 002DD8 700021D8 27A50060 */ addiu $a1, $sp, 0x60
/* 002DDC 700021DC 0C003774 */ jal osRecvMesg
/* 002DE0 700021E0 03C03025 */ move $a2, $fp
/* 002DE4 700021E4 0C0008F9 */ jal amHandleDoneMessage
/* 002DE8 700021E8 8FA40060 */ lw $a0, 0x60($sp)
/* 002DEC 700021EC 10000004 */ b .L70002200
/* 002DF0 700021F0 00000000 */ nop
.L700021F4:
/* 002DF4 700021F4 10000002 */ b .L70002200
/* 002DF8 700021F8 03C09025 */ move $s2, $fp
/* 002DFC 700021FC 03C09025 */ move $s2, $fp
.L70002200:
/* 002E00 70002200 5240FF82 */ beql $s2, $zero, .L7000200C
/* 002E04 70002204 02C02025 */ move $a0, $s6
/* 002E08 70002208 3C048006 */ lui $a0, %hi(g_AudioManager+0x238)
/* 002E0C 7000220C 0C003AB9 */ jal alClose
/* 002E10 70002210 2484E750 */ addiu $a0, %lo(g_AudioManager+0x238) # addiu $a0, $a0, -0x18b0
/* 002E14 70002214 8FBF003C */ lw $ra, 0x3c($sp)
/* 002E18 70002218 8FB00018 */ lw $s0, 0x18($sp)
/* 002E1C 7000221C 8FB1001C */ lw $s1, 0x1c($sp)
/* 002E20 70002220 8FB20020 */ lw $s2, 0x20($sp)
/* 002E24 70002224 8FB30024 */ lw $s3, 0x24($sp)
/* 002E28 70002228 8FB40028 */ lw $s4, 0x28($sp)
/* 002E2C 7000222C 8FB5002C */ lw $s5, 0x2c($sp)
/* 002E30 70002230 8FB60030 */ lw $s6, 0x30($sp)
/* 002E34 70002234 8FB70034 */ lw $s7, 0x34($sp)
/* 002E38 70002238 8FBE0038 */ lw $fp, 0x38($sp)
/* 002E3C 7000223C 03E00008 */ jr $ra
/* 002E40 70002240 27BD0070 */ addiu $sp, $sp, 0x70
)
#endif
/**
* 2E44 70002244
* Based on method
* static u32 __amHandleFrameMsg(AudioInfo *info, AudioInfo *lastInfo)
* from the n64devkit demos_old/simple/audiomgr.c.
*
* original documentation:
* First, clear the past audio dma's, then calculate
* the number of samples you will need for this frame. This value varies
* due to the fact that audio is synchronised off of the video interupt
* which can have a small amount of jitter in it. Varying the number of
* samples slightly will allow you to stay in synch with the video. This
* is an advantageous thing to do, since if you are in synch with the
* video, you will have fewer graphics yields. After you've calculated
* the number of frames needed, call alAudioFrame, which will call all
* of the synthesizer's players (sequence player and sound player) to
* generate the audio task list. If you get a valid task list back, put
* it in a task structure and send a message to the scheduler to let it
* know that the next frame of audio is ready for processing.
*
* @param info audio info.
* @param lastInfo last info.
*/
void amHandleFrameMessage(AudioInfo *info, AudioInfo *lastInfo)
{
s16* outBuffer;
Acmd *cmdlp;
s32 temp_v1;
/* call once a frame, before doing alAudioFrame */
amClearDmaBuffers();
outBuffer = (s16*)osVirtualToPhysical(info->data);
if (lastInfo)
{
osAiSetNextBuffer(lastInfo->data, lastInfo->frameSamples * 4);
}
/* calculate how many samples needed for this frame to keep the DAC full */
/* this will vary slightly frame to frame, must recalculate every frame */
/* divide by four, to convert bytes */
/* to stereo 16 bit samples */
info->frameSamples = (u16)(((g_FrameSize - (osAiGetLength() >> 2)) + 16 + EXTRA_SAMPLES) & ~0xf);
temp_v1 = g_MinFrameSize;
if ((s32)info->frameSamples < (s32)(s16)temp_v1)
{
info->frameSamples = (s16)temp_v1;
}
cmdlp = (Acmd*)alAudioFrame(g_AudioManager.cmdList[g_CurrentAcmdList], &g_CommandLength, outBuffer, info->frameSamples);
/* paranoia */
info->task.next = 0;
info->task.flags = 0;
/* reply to when finished */
info->task.msgQ = (void *) (&(g_AudioManager.replyMessageQueue.mtqueue));
/* reply with this message */
info->task.msg = info;
info->task.flags = OS_SC_NEEDS_RSP;
info->task.list.t.data_ptr = (u64*)(g_AudioManager.cmdList[g_CurrentAcmdList]);
info->task.list.t.data_size = (((s32)cmdlp - (s32)g_AudioManager.cmdList[g_CurrentAcmdList]) >> 3) * sizeof(Acmd);
info->task.list.t.type = M_AUDTASK;
info->task.list.t.ucode_boot = (u64*)rspbootTextStart;
info->task.list.t.ucode_boot_size = ((s32)gsp3DTextStart - (s32)rspbootTextStart);
info->task.list.t.flags = 0; // 1c
info->task.list.t.ucode = (u64*)aspMainTextStart;
info->task.list.t.ucode_data = (u64*)aspMainDataStart;
info->task.list.t.ucode_data_size = SP_UCODE_DATA_SIZE;
info->task.list.t.yield_data_ptr = NULL; // 50
info->task.list.t.yield_data_size = 0; // 54
osSendMesg(osScGetCmdQ(&sc), (OSMesg)&info->task, OS_MESG_NOBLOCK);
/* swap which acmd list you use each frame */
g_CurrentAcmdList ^= 1;
}
/**
* 2FE4 700023E4
* Based on method
* static void __amHandleDoneMsg(AudioInfo *info)
* from the n64devkit demos_old/simple/audiomgr.c.
*
* original documentation:
* Really just debugging info in this frame. Checks
* to make sure we completed before we were out of samples.
*
* @param info Unused.
*/
void amHandleDoneMessage(AudioInfo *info)
{
s32 samplesLeft;
/*
* in the audiomgr example, firstTime is declared here with
* the static keyword. That breaks the build, but the following
* code will compile to a matching binary,
*/
int *b;
samplesLeft = (s32)osAiGetLength() >> 2;
/*
* The initial code probably looked like the following (and this
* is what you get with mips_to_c):
*
* if (samplesLeft == 0 && !firstTime)
*/
b = &g_FirstTime;
if (!samplesLeft && !(*b))
{
// debug printf from audioMgr demo
#ifdef DEBUG
osSyncPrintf("audio: ai out of samples\n");
#endif
g_FirstTime = 0;
}
}
/**
* 3024 70002424
* Looks to be based on method
* s32 __amDMA(s32 addr, s32 len, void *state)
* from the n64devkit.
*
* original documentation:
* This routine handles the dma'ing of samples from rom to ram.
* First it checks the current buffers to see if the samples needed are
* already in place. Because buffers are linked sequentially by the
* addresses where the samples are on rom, it doesn't need to check all
* of them, only up to the address that it needs. If it finds one, it
* returns the address of that buffer. If it doesn't find the samples
* that it needs, it will initiate a DMA of the samples that it needs.
* In either case, it updates the lastFrame variable, to indicate that
* this buffer was last used in this frame. This is important for the
* __clearAudioDMA routine.
*
* @param addr ?.
* @param len ?.
* @param state unused.
* @return result from call to osVirtualToPhysical
*/
s32 amDmaCallback(s32 addr, s32 len, void* state)
{
void *freeBuffer;
s32 delta;
DMABuffer *dmaPtr;
s32 addrEnd;
s32 buffEnd;
DMABuffer *lastDmaPtr;
lastDmaPtr = NULL;
dmaPtr = g_DmaState.firstUsed;
delta = addr & 0x1;
addrEnd = addr + len;
/* first check to see if a currently existing buffer contains the
sample that you need. */
while (dmaPtr)
{
buffEnd = dmaPtr->startAddr + AUDIO_DMA_MAX_BUFFER_LENGTH;
/* since buffers are ordered */
/* abort if past possible */
if ((u32)dmaPtr->startAddr > (u32)addr)
{
break;
}
/* yes, found a buffer with samples */
else if (addrEnd <= buffEnd)
{
/* mark it used */
dmaPtr->lastFrame = (s32) g_AudioFrameCount;
freeBuffer = (dmaPtr->ptr + addr) - dmaPtr->startAddr;
return osVirtualToPhysical(freeBuffer);
}
lastDmaPtr = dmaPtr;
dmaPtr = (DMABuffer*)dmaPtr->node.next;
}
/* get here, and you didn't find a buffer, so dma a new one */
/* get a buffer from the free list */
dmaPtr = g_DmaState.firstFree;
/*
* if you get here and dmaPtr is null, send back a bogus
* pointer, it's better than nothing
*/
if (!dmaPtr)
{
if (!lastDmaPtr)
{
lastDmaPtr = g_DmaState.firstUsed;
}
return osVirtualToPhysical(lastDmaPtr->ptr) + delta;
}
g_DmaState.firstFree = (DMABuffer*)dmaPtr->node.next;
alUnlink((ALLink*)dmaPtr);
/* add it to the used list */
/* if you have other dmabuffers used, add this one */
/* to the list, after the last one checked above */
if (lastDmaPtr)
{
alLink((ALLink*)dmaPtr, (ALLink*)lastDmaPtr);
}
/* if this buffer is before any others */
// Jam at begining of list
else if (g_DmaState.firstUsed)
{
lastDmaPtr = g_DmaState.firstUsed;
g_DmaState.firstUsed = dmaPtr;
dmaPtr->node.next = (ALLink*)lastDmaPtr;
dmaPtr->node.prev = 0;
lastDmaPtr->node.prev = (ALLink*)dmaPtr;
}
/* no buffers in list, this is the first one */
else
{
g_DmaState.firstUsed = dmaPtr;
dmaPtr->node.next = 0;
dmaPtr->node.prev = 0;
}
freeBuffer = dmaPtr->ptr;
addr -= delta;
dmaPtr->startAddr = addr;
dmaPtr->lastFrame = g_AudioFrameCount;
osPiStartDma(&g_DmaIOMessageBuffer[g_NextDMa++], OS_MESG_PRI_HIGH, OS_READ, (u32)addr, freeBuffer, AUDIO_DMA_MAX_BUFFER_LENGTH, &g_DmaMessageQueue);
return (s32)osVirtualToPhysical(freeBuffer) + delta;
}
/**
* 31D8 700025D8
* Based on method
* ALDMAproc __amDmaNew(AMDMAState **state)
* from the n64devkit demos_old/simple/audiomgr.c.
*
* original documentation:
* Initialize the dma buffers and return the address of the
* procedure that will be used to dma the samples from rom to ram. This
* routine will be called once for each physical voice that is created.
* In this case, because we know where all the buffers are, and since
* they are not attached to a specific voice, we will only really do any
* initialization the first time. After that we just return the address
* to the dma routine.
*
* @param state will point to g_DmaState after call.
* @return Address of dma callback function.
*/
ALDMAproc amDmaNew(DMAState** state)
{
if (g_DmaState.u.initialized == 0)
{
g_DmaState.firstUsed = NULL;
g_DmaState.firstFree = g_DmaBuffers;
g_DmaState.u.initialized = (u8)1U;
}
*state = &g_DmaState;
return &amDmaCallback;
}
/**
* 3210 70002610
* Based on method
* static void __clearAudioDMA(void)
* from the n64devkit demos_old/simple/audiomgr.c.
*
* original documentation:
* Routine to move dma buffers back to the unused list.
* First clear out your dma messageQ. Then check each buffer to see when
* it was last used. If that was more than FRAME_LAG frames ago, move it
* back to the unused list.
*/
void amClearDmaBuffers(void)
{
u32 i;
OSMesg osmesg;
DMABuffer *dmaPtr, *nextPtr;
osmesg = 0;
/*
* Don't block here. If dma's aren't complete, you've had an audio
* overrun. (Bad news, but go for it anyway, and try and recover.
*/
for (i=0; i < g_NextDMa; i++)
{
if (osRecvMesg(&g_DmaMessageQueue, (OSMesg *)&osmesg, OS_MESG_NOBLOCK) == -1)
{
#ifdef DEBUG
osSyncPrintf("Dma not done\n");
#endif
}
#ifdef DEBUG
/* debug logging from audioMgr.c, I think this requires #include <ultralog.h>
* // if (logging)
* // osLogEvent(log, 17, 2, osmesg->devAddr, osmesg->size);
*/
#endif
}
dmaPtr = g_DmaState.firstUsed;
while (dmaPtr)
{
nextPtr = (DMABuffer*)dmaPtr->node.next;
/* remove old dma's from list */
/* Can change FRAME_LAG value. Should be at least one. */
/* Larger values mean more buffers needed, but fewer DMA's */
if (dmaPtr->lastFrame + 1 < g_AudioFrameCount)
{
if (g_DmaState.firstUsed == dmaPtr)
{
g_DmaState.firstUsed = (DMABuffer*)dmaPtr->node.next;
}
alUnlink((ALLink*)dmaPtr);
if (g_DmaState.firstFree)
{
alLink((ALLink*)dmaPtr, (ALLink*)g_DmaState.firstFree);
}
else
{
g_DmaState.firstFree = dmaPtr;
dmaPtr->node.next = 0;
dmaPtr->node.prev = 0;
}
}
dmaPtr = nextPtr;
}
g_NextDMa = 0U;
g_AudioFrameCount = (s32)(g_AudioFrameCount + 1);
}
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