properly merge refresh 10.1's audio code

this fixes crashes on EU, there is now audio output but it's still borked
This commit is contained in:
fgsfds 2020-06-21 02:22:37 +03:00
parent 8fa922c5e0
commit 531a483234
8 changed files with 902 additions and 551 deletions

View File

@ -763,10 +763,6 @@ void func_eu_802e9bec(s32 player, s32 channel, s32 arg2) {
}
#else
// Stubbed N64-US/JP audio code
// continue;
#endif
struct SPTask *create_next_audio_frame_task(void) {
return NULL;
@ -783,6 +779,7 @@ void create_next_audio_buffer(s16 *samples, u32 num_samples) {
gAudioRandom = ((gAudioRandom + gAudioFrameCount) * gAudioFrameCount);
decrease_sample_dma_ttls();
}
#endif
void play_sound(s32 soundBits, f32 *pos) {
sSoundRequests[sSoundRequestCount].soundBits = soundBits;

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@ -648,9 +648,8 @@ s32 audio_shut_down_and_reset_step(void) {
/**
* Waits until a specified number of audio frames have been created
*/
void wait_for_audio_frames(s32 frames) {
void wait_for_audio_frames(UNUSED s32 frames) {
gAudioFrameCount = 0;
}
#endif

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@ -894,13 +894,13 @@ void audio_init() {
UNUSED s8 pad[32];
u8 buf[0x10];
#endif
s32 i, j, k;
s32 i, j, UNUSED k;
UNUSED s32 lim1; // lim1 unused in EU
#ifdef VERSION_EU
u8 buf[0x10];
s32 UNUSED lim2, lim3;
#else
s32 lim2, lim3;
s32 lim2, UNUSED lim3;
#endif
u32 size;
UNUSED u64 *ptr64;
@ -920,7 +920,6 @@ void audio_init() {
for (i = 0; i <= lim2 / 8 - 1; i++) {
((u64 *) gAudioHeap)[i] = 0;
}
#else
for (i = 0; i < gAudioHeapSize / 8; i++) {
((u64 *) gAudioHeap)[i] = 0;

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@ -35,100 +35,28 @@ s32 audio_shut_down_and_reset_step(void);
void func_802ad7ec(u32);
struct SPTask *create_next_audio_frame_task(void) {
u32 samplesRemainingInAI;
return NULL;
}
void create_next_audio_buffer(s16 *samples, u32 num_samples) {
s32 writtenCmds;
s32 index;
OSTask_t *task;
s32 flags;
s16 *currAiBuffer;
s32 oldDmaCount;
OSMesg sp30;
OSMesg sp2C;
OSMesg msg;
gAudioFrameCount++;
if (gAudioFrameCount % gAudioBufferParameters.presetUnk4 != 0) {
stubbed_printf("DAC:Lost 1 Frame.\n");
return NULL;
}
osSendMesg(OSMesgQueues[0], (OSMesg) gAudioFrameCount, 0);
gAudioTaskIndex ^= 1;
gCurrAiBufferIndex++;
gCurrAiBufferIndex %= NUMAIBUFFERS;
index = (gCurrAiBufferIndex - 2 + NUMAIBUFFERS) % NUMAIBUFFERS;
samplesRemainingInAI = osAiGetLength() / 4;
if (gAiBufferLengths[index] != 0) {
osAiSetNextBuffer(gAiBuffers[index], gAiBufferLengths[index] * 4);
}
oldDmaCount = gCurrAudioFrameDmaCount;
if (oldDmaCount > AUDIO_FRAME_DMA_QUEUE_SIZE) {
stubbed_printf("DMA: Request queue over.( %d )\n", oldDmaCount);
}
gCurrAudioFrameDmaCount = 0;
decrease_sample_dma_ttls();
if (osRecvMesg(OSMesgQueues[2], &sp30, 0) != -1) {
gAudioResetPresetIdToLoad = (u8) (s32) sp30;
if (osRecvMesg(OSMesgQueues[2], &msg, 0) != -1) {
gAudioResetPresetIdToLoad = (u8) (s32) msg;
gAudioResetStatus = 5;
}
if (gAudioResetStatus != 0) {
if (audio_shut_down_and_reset_step() == 0) {
if (gAudioResetStatus == 0) {
osSendMesg(OSMesgQueues[3], (OSMesg) (s32) gAudioResetPresetIdToLoad, OS_MESG_NOBLOCK);
}
return NULL;
}
audio_reset_session();
gAudioResetStatus = 0;
}
gAudioTask = &gAudioTasks[gAudioTaskIndex];
gAudioCmd = gAudioCmdBuffers[gAudioTaskIndex];
index = gCurrAiBufferIndex;
currAiBuffer = gAiBuffers[index];
gAiBufferLengths[index] = ((gAudioBufferParameters.samplesPerFrameTarget - samplesRemainingInAI +
EXTRA_BUFFERED_AI_SAMPLES_TARGET) & ~0xf) + SAMPLES_TO_OVERPRODUCE;
if (gAiBufferLengths[index] < gAudioBufferParameters.minAiBufferLength) {
gAiBufferLengths[index] = gAudioBufferParameters.minAiBufferLength;
if (osRecvMesg(OSMesgQueues[1], &msg, OS_MESG_NOBLOCK) != -1) {
func_802ad7ec((u32) msg);
}
if (gAiBufferLengths[index] > gAudioBufferParameters.maxAiBufferLength) {
gAiBufferLengths[index] = gAudioBufferParameters.maxAiBufferLength;
}
if (osRecvMesg(OSMesgQueues[1], &sp2C, OS_MESG_NOBLOCK) != -1) {
func_802ad7ec((u32) sp2C);
}
flags = 0;
gAudioCmd = synthesis_execute(gAudioCmd, &writtenCmds, currAiBuffer, gAiBufferLengths[index]);
synthesis_execute(gAudioCmdBuffers[0], &writtenCmds, samples, num_samples);
gAudioRandom = ((gAudioRandom + gAudioFrameCount) * gAudioFrameCount);
gAudioRandom = gAudioRandom + writtenCmds / 8;
index = gAudioTaskIndex;
gAudioTask->msgqueue = NULL;
gAudioTask->msg = NULL;
task = &gAudioTask->task.t;
task->type = M_AUDTASK;
task->flags = flags;
task->ucode_boot = rspF3DBootStart;
task->ucode_boot_size = (u8 *) rspF3DBootEnd - (u8 *) rspF3DBootStart;
task->ucode = rspAspMainStart;
task->ucode_data = rspAspMainDataStart;
task->ucode_size = 0x800; // (this size is ignored)
task->ucode_data_size = (rspAspMainDataEnd - rspAspMainDataStart) * sizeof(u64);
task->dram_stack = NULL;
task->dram_stack_size = 0;
task->output_buff = NULL;
task->output_buff_size = NULL;
task->data_ptr = gAudioCmdBuffers[index];
task->data_size = writtenCmds * sizeof(u64);
task->yield_data_ptr = NULL;
task->yield_data_size = 0;
return gAudioTask;
}
void eu_process_audio_cmd(struct EuAudioCmd *cmd) {

File diff suppressed because it is too large Load Diff

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@ -1,7 +1,8 @@
#ifndef MIXER_H
#define MIXER_H
#include <PR/ultratypes.h>
#include <stdint.h>
#include <ultra64.h>
#undef aSegment
#undef aClearBuffer
@ -19,21 +20,34 @@
#undef aLoadADPCM
#undef aADPCMdec
#define aSegment(pkt, s, b)
void aClearBuffer(uint64_t *cmd, uint16_t dmem, uint16_t count);
void aSetBuffer(uint64_t *cmd, uint8_t flags, uint16_t dmemin, uint16_t dmemout, uint16_t count);
void aLoadBuffer(uint64_t *cmd, uint16_t *addr);
void aSaveBuffer(uint64_t *cmd, uint16_t *addr);
void aDMEMMove(uint64_t *cmd, uint16_t dmemin, uint16_t dmemout, uint16_t count);
void aMix(uint64_t *cmd, uint8_t flags, uint16_t gain, uint16_t dmemin, uint16_t dmemout);
void aEnvMixer(uint64_t *cmd, uint8_t flags, uint16_t *addr);
void aResample(uint64_t *cmd, uint8_t flags, uint16_t pitch, uint16_t *state_addr);
void aInterleave(uint64_t *cmd, uint16_t inL, uint16_t inR);
void aSetVolume(uint64_t *cmd, uint8_t flags, uint16_t vol, uint16_t voltgt, uint16_t volrate);
void aSetVolume32(uint64_t *cmd, uint8_t flags, uint16_t voltgt, uint32_t volrate);
void aSetLoop(uint64_t *cmd, uint16_t *addr);
void aLoadADPCM(uint64_t *cmd, uint16_t count, uint16_t *addr);
void aADPCMdec(uint64_t *cmd, uint8_t flags, uint16_t *last_frame_addr);
void aClearBufferImpl(uint16_t addr, int nbytes);
void aLoadBufferImpl(const void *source_addr);
void aSaveBufferImpl(int16_t *dest_addr);
void aLoadADPCMImpl(int num_entries_times_16, const int16_t *book_source_addr);
void aSetBufferImpl(uint8_t flags, uint16_t in, uint16_t out, uint16_t nbytes);
void aSetVolumeImpl(uint8_t flags, int16_t v, int16_t t, int16_t r);
void aInterleaveImpl(uint16_t left, uint16_t right);
void aDMEMMoveImpl(uint16_t in_addr, uint16_t out_addr, int nbytes);
void aSetLoopImpl(ADPCM_STATE *adpcm_loop_state);
void aADPCMdecImpl(uint8_t flags, ADPCM_STATE state);
void aResampleImpl(uint8_t flags, uint16_t pitch, RESAMPLE_STATE state);
void aEnvMixerImpl(uint8_t flags, ENVMIX_STATE state);
void aMixImpl(int16_t gain, uint16_t in_addr, uint16_t out_addr);
#define aSegment(pkt, s, b) do { } while(0)
#define aClearBuffer(pkt, d, c) aClearBufferImpl(d, c)
#define aLoadBuffer(pkt, s) aLoadBufferImpl(s)
#define aSaveBuffer(pkt, s) aSaveBufferImpl(s)
#define aLoadADPCM(pkt, c, d) aLoadADPCMImpl(c, d)
#define aSetBuffer(pkt, f, i, o, c) aSetBufferImpl(f, i, o, c)
#define aSetVolume(pkt, f, v, t, r) aSetVolumeImpl(f, v, t, r)
#define aSetVolume32(pkt, f, v, tr) aSetVolume(pkt, f, v, (int16_t)((tr) >> 16), (int16_t)(tr))
#define aInterleave(pkt, l, r) aInterleaveImpl(l, r)
#define aDMEMMove(pkt, i, o, c) aDMEMMoveImpl(i, o, c)
#define aSetLoop(pkt, a) aSetLoopImpl(a)
#define aADPCMdec(pkt, f, s) aADPCMdecImpl(f, s)
#define aResample(pkt, f, p, s) aResampleImpl(f, p, s)
#define aEnvMixer(pkt, f, s) aEnvMixerImpl(f, s)
#define aMix(pkt, f, g, i, o) aMixImpl(g, i, o)
#endif

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@ -72,7 +72,13 @@ void send_display_list(struct SPTask *spTask) {
gfx_run((Gfx *)spTask->task.t.data_ptr);
}
#define printf
#ifdef VERSION_EU
#define SAMPLES_HIGH 656
#define SAMPLES_LOW 640
#else
#define SAMPLES_HIGH 544
#define SAMPLES_LOW 528
#endif
void produce_one_frame(void) {
gfx_start_frame();
@ -86,9 +92,9 @@ void produce_one_frame(void) {
thread6_rumble_loop(NULL);
int samples_left = audio_api->buffered();
u32 num_audio_samples = samples_left < audio_api->get_desired_buffered() ? 544 : 528;
u32 num_audio_samples = samples_left < audio_api->get_desired_buffered() ? SAMPLES_HIGH : SAMPLES_LOW;
//printf("Audio samples: %d %u\n", samples_left, num_audio_samples);
s16 audio_buffer[544 * 2 * 2];
s16 audio_buffer[SAMPLES_HIGH * 2 * 2];
for (int i = 0; i < 2; i++) {
/*if (audio_cnt-- == 0) {
audio_cnt = 2;
@ -98,7 +104,7 @@ void produce_one_frame(void) {
}
//printf("Audio samples before submitting: %d\n", audio_api->buffered());
audio_api->play((u8*)audio_buffer, 2 * num_audio_samples * 4);
audio_api->play((u8 *)audio_buffer, 2 * num_audio_samples * 4);
gfx_end_frame();
}

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@ -22,15 +22,42 @@ s32 osPiStartDma(UNUSED OSIoMesg *mb, UNUSED s32 priority, UNUSED s32 direction,
return 0;
}
void osCreateMesgQueue(OSMesgQueue *mq, OSMesg *msgBuf, s32 count) {
mq->validCount = 0;
mq->first = 0;
mq->msgCount = count;
mq->msg = msgBuf;
return;
}
void osSetEventMesg(UNUSED OSEvent e, UNUSED OSMesgQueue *mq, UNUSED OSMesg msg) {
}
s32 osJamMesg(UNUSED OSMesgQueue *mq, UNUSED OSMesg msg, UNUSED s32 flag) {
return 0;
}
s32 osSendMesg(UNUSED OSMesgQueue *mq, UNUSED OSMesg msg, UNUSED s32 flag) {
#ifdef VERSION_EU
s32 index;
if (mq->validCount >= mq->msgCount) {
return -1;
}
index = (mq->first + mq->validCount) % mq->msgCount;
mq->msg[index] = msg;
mq->validCount++;
#endif
return 0;
}
s32 osRecvMesg(UNUSED OSMesgQueue *mq, UNUSED OSMesg *msg, UNUSED s32 flag) {
#ifdef VERSION_EU
if (mq->validCount == 0) {
return -1;
}
if (msg != NULL) {
*msg = *(mq->first + mq->msg);
}
mq->first = (mq->first + 1) % mq->msgCount;
mq->validCount--;
#endif
return 0;
}