I tried with pole
Build libsm64 / Build libsm64 for linux (push) Successful in 31s
Build libsm64 / Build libsm64 for windows (push) Successful in 23s
Build libsm64 / Snapshot Release (push) Has been cancelled
Build libsm64 / Build libsm64 for windows (pull_request) Has been cancelled
Build libsm64 / Snapshot Release (pull_request) Has been cancelled
Build libsm64 / Build libsm64 for linux (pull_request) Has been cancelled

This commit is contained in:
2026-06-01 23:29:58 -05:00
parent 53731adf81
commit 8daff2b6a2
15 changed files with 89 additions and 290 deletions
-4
View File
@@ -872,14 +872,12 @@ static void process_sound_request(u32 bits, f32 *pos)
if (soundId >= sNumSoundsPerBank[bank] || sSoundBankDisabled[bank])
{
//DEBUG_PRINT("process_sound_request: invalid soundId %d\n", soundId);
return;
}
u8 soundIndex = sSoundBanks[bank][0].next;
while (soundIndex != 0xff && soundIndex != 0)
{
//DEBUG_PRINT("process_sound_request: soundIndex %d\n", soundIndex);
// If an existing sound from the same source exists in the bank, then we should either
// interrupt that sound and replace it with the new sound, or we should drop the new sound.
if (sSoundBanks[bank][soundIndex].x == pos)
@@ -930,7 +928,6 @@ static void process_sound_request(u32 bits, f32 *pos)
// If free list has more than one element remaining
if (sSoundBanks[bank][sSoundBankFreeListFront[bank]].next != 0xff && soundIndex != 0)
{
DEBUG_PRINT("process_sound_request2: soundIndex %d\n", soundIndex);
// Allocate from free list
soundIndex = sSoundBankFreeListFront[bank];
@@ -966,7 +963,6 @@ static void process_all_sound_requests(void)
struct Sound *sound = &sSoundRequests[sNumProcessedSoundRequests];
process_sound_request(sound->soundBits, sound->position);
sNumProcessedSoundRequests++;
//DEBUG_PRINT("processed sounds: %d\n", sNumProcessedSoundRequests);
}
}
-51
View File
@@ -120,17 +120,10 @@ ALSeqFile *get_audio_file_header(s32 arg0);
*/
void audio_dma_copy_immediate(uintptr_t devAddr, void *vAddr, size_t nbytes)
{
DEBUG_PRINT("audio_dma_copy_immediate()");
DEBUG_PRINT("- dev addr: %x", devAddr);
DEBUG_PRINT("- vAddr: %x", vAddr);
DEBUG_PRINT("- # bytes: %d", nbytes);
eu_stubbed_printf_3("Romcopy %x -> %x ,size %x\n", devAddr, vAddr, nbytes);
DEBUG_PRINT("- invalidate d cache");
osInvalDCache(vAddr, nbytes);
DEBUG_PRINT("- start dma");
osPiStartDma(&gAudioDmaIoMesg, OS_MESG_PRI_HIGH, OS_READ, devAddr, vAddr, nbytes,
&gAudioDmaMesgQueue);
DEBUG_PRINT("- recv message");
osRecvMesg(&gAudioDmaMesgQueue, NULL, OS_MESG_BLOCK);
eu_stubbed_printf_0("Romcopyend\n");
}
@@ -646,43 +639,31 @@ void patch_audio_bank(struct AudioBank *mem, u8 *offset, u32 numInstruments, u32
struct AudioBank *bank_load_immediate(s32 bankId, s32 arg1)
{
DEBUG_PRINT("bank_load_immediate()");
UNUSED u32 pad1[4];
u32 buf[4];
// (This is broken if the length is 1 (mod 16), but that never happens --
// it's always divisible by 4.)
DEBUG_PRINT("- getting alloc");
s32 alloc = gAlCtlHeader->seqArray[bankId].len + 0xf;
DEBUG_PRINT("- aligning");
alloc = ALIGN16(alloc);
alloc -= 0x10;
DEBUG_PRINT("- getting ctl data for bank %d", bankId);
u8 *ctlData = gAlCtlHeader->seqArray[bankId].offset;
DEBUG_PRINT("- alloc bank or seq");
struct AudioBank *ret = alloc_bank_or_seq(&gBankLoadedPool, 1, alloc, arg1, bankId);
if (ret == NULL)
{
return NULL;
}
DEBUG_PRINT("- copying dma immediate 1");
DEBUG_PRINT("- ctlData: %x", ctlData);
audio_dma_copy_immediate((uintptr_t)ctlData, buf, 0x10);
DEBUG_PRINT("- getting nums");
u32 numInstruments = buf[0];
u32 numDrums = buf[1];
DEBUG_PRINT("- copying dma immediate 2");
audio_dma_copy_immediate((uintptr_t)(ctlData + 0x10), ret, alloc);
DEBUG_PRINT("- patching bank");
patch_audio_bank(ret, gAlTbl->seqArray[bankId].offset, numInstruments, numDrums);
DEBUG_PRINT("- setting ctl entries");
gCtlEntries[bankId].numInstruments = (u8)numInstruments;
gCtlEntries[bankId].numDrums = (u8)numDrums;
gCtlEntries[bankId].instruments = ret->instruments;
gCtlEntries[bankId].drums = ret->drums;
DEBUG_PRINT("- setting load status");
gBankLoadStatus[bankId] = SOUND_LOAD_STATUS_COMPLETE;
return ret;
}
@@ -851,12 +832,10 @@ u8 get_missing_bank(u32 seqId, s32 *nonNullCount, s32 *nullCount)
struct AudioBank *load_banks_immediate(s32 seqId, u8 *outDefaultBank)
{
DEBUG_PRINT("load_banks_immediate()");
void *ret;
u32 bankId;
u8 i;
DEBUG_PRINT("- getting offset");
u16 offset = ((u16 *)gAlBankSets)[seqId];
#ifdef VERSION_EU
for (i = gAlBankSets[offset++]; i != 0; i--)
@@ -864,21 +843,17 @@ struct AudioBank *load_banks_immediate(s32 seqId, u8 *outDefaultBank)
bankId = gAlBankSets[offset++];
#else
offset++;
DEBUG_PRINT("- looping through bank sets");
for (i = gAlBankSets[offset - 1]; i != 0; i--)
{
offset++;
DEBUG_PRINT("- getting bank id");
bankId = gAlBankSets[offset - 1];
#endif
DEBUG_PRINT("- checking if bank load is complete");
if (IS_BANK_LOAD_COMPLETE(bankId) == TRUE)
{
#ifdef VERSION_EU
ret = get_bank_or_seq(&gBankLoadedPool, 2, bankId);
#else
DEBUG_PRINT("- getting bank or seq");
ret = get_bank_or_seq(&gBankLoadedPool, 2, gAlBankSets[offset - 1]);
#endif
}
@@ -889,7 +864,6 @@ struct AudioBank *load_banks_immediate(s32 seqId, u8 *outDefaultBank)
if (ret == NULL)
{
DEBUG_PRINT("- bank load immediate");
ret = bank_load_immediate(bankId, 2);
}
}
@@ -899,7 +873,6 @@ struct AudioBank *load_banks_immediate(s32 seqId, u8 *outDefaultBank)
void preload_sequence(u32 seqId, u8 preloadMask)
{
DEBUG_PRINT("preload_sequence()");
void *sequenceData;
u8 temp;
@@ -911,25 +884,21 @@ void preload_sequence(u32 seqId, u8 preloadMask)
gAudioLoadLock = AUDIO_LOCK_LOADING;
if (preloadMask & PRELOAD_BANKS)
{
DEBUG_PRINT("- load banks immediate");
load_banks_immediate(seqId, &temp);
}
if (preloadMask & PRELOAD_SEQUENCE)
{
// @bug should be IS_SEQ_LOAD_COMPLETE
DEBUG_PRINT("- checking if bank load immediate");
if (IS_BANK_LOAD_COMPLETE(seqId) == TRUE)
{
eu_stubbed_printf_1("SEQ %d ALREADY CACHED\n", seqId);
DEBUG_PRINT("- getting bank or seq");
sequenceData = get_bank_or_seq(&gSeqLoadedPool, 2, seqId);
}
else
{
sequenceData = NULL;
}
DEBUG_PRINT("- checking dma immediate");
if (sequenceData == NULL && sequence_dma_immediate(seqId, 2) == NULL)
{
gAudioLoadLock = AUDIO_LOCK_NOT_LOADING;
@@ -1037,7 +1006,6 @@ void load_sequence_internal(u32 player, u32 seqId, s32 loadAsync)
// (void) must be omitted from parameters to fix stack with -framepointer
void audio_init()
{
DEBUG_PRINT("audio_init()");
UNUSED s8 pad[32];
u8 buf[0x10];
s32 i, UNUSED k;
@@ -1048,7 +1016,6 @@ void audio_init()
gAudioLoadLock = AUDIO_LOCK_UNINITIALIZED;
DEBUG_PRINT("- setting values in unused");
s32 lim1 = gUnusedCount80333EE8;
for (i = 0; i < lim1; i++)
{
@@ -1056,7 +1023,6 @@ void audio_init()
gUnused80226E98[i] = 0;
}
DEBUG_PRINT("- clearing audio heap");
s32 lim2 = gAudioHeapSize;
for (i = 0; i <= lim2 / 8 - 1; i++)
{
@@ -1114,39 +1080,22 @@ void audio_init()
alSeqFileNew(gSeqFileHeader, data);
// Load header for CTL (instrument metadata)
DEBUG_PRINT("- loading ctl header");
gAlCtlHeader = (ALSeqFile *)buf;
data = gSoundDataADSR;
DEBUG_PRINT("- copying dma immediate");
audio_dma_copy_immediate((uintptr_t)data, gAlCtlHeader, 0x10);
size = gAlCtlHeader->seqCount * sizeof(ALSeqData) + 4;
DEBUG_PRINT("- seq count: %d", gAlCtlHeader->seqCount);
DEBUG_PRINT("- size after read: %d", size);
size = ALIGN16(size);
DEBUG_PRINT("- size after align: %d", size);
gCtlEntries = soundAlloc(&gAudioInitPool, gAlCtlHeader->seqCount * sizeof(struct CtlEntry));
gAlCtlHeader = soundAlloc(&gAudioInitPool, size);
DEBUG_PRINT("@ copying data from sound data adsr to ctl header");
DEBUG_PRINT("- data: %x", data);
DEBUG_PRINT("- ctl header: %x", gAlCtlHeader);
DEBUG_PRINT("- size: %d", size);
audio_dma_copy_immediate((uintptr_t)data, gAlCtlHeader, size);
DEBUG_PRINT("- creating new seq file for ctl");
alSeqFileNew(gAlCtlHeader, data);
// Load header for TBL (raw sound data)
DEBUG_PRINT("- loading tbl");
gAlTbl = (ALSeqFile *)buf;
DEBUG_PRINT("- copying dma");
audio_dma_copy_immediate((uintptr_t)data, gAlTbl, 0x10);
DEBUG_PRINT("- tbl seq count: %d", gAlTbl->seqCount);
size = gAlTbl->seqCount * sizeof(ALSeqData) + 4;
DEBUG_PRINT("- size: %d", size);
size = ALIGN16(size);
DEBUG_PRINT("- size after align: %d", size);
gAlTbl = soundAlloc(&gAudioInitPool, size);
DEBUG_PRINT("- tbl alloc at %x", gAlTbl);
DEBUG_PRINT("- gSoundDataRaw at %x", gSoundDataRaw);
audio_dma_copy_immediate((uintptr_t)gSoundDataRaw, gAlTbl, size);
alSeqFileNew(gAlTbl, (u8 *)gSoundDataRaw);
-20
View File
@@ -447,10 +447,7 @@ void init_layer_freelist(void)
u8 m64_read_u8(struct M64ScriptState *state)
{
DEBUG_PRINT("m64_read_u8()");
DEBUG_PRINT("- state at %x", state);
u8 *midiArg = state->pc++;
DEBUG_PRINT("- read u8 (%d) at (%x)", *midiArg, midiArg);
return *midiArg;
}
@@ -1205,10 +1202,8 @@ s32 seq_channel_layer_process_script_part2(struct SequenceChannelLayer *layer)
break;
case 0xc6: // layer_setinstr
DEBUG_PRINT(" - cmd 0xc6, setinstr");
cmd = m64_read_u8(state);
DEBUG_PRINT(" - read %d from state", cmd);
if (cmd >= 0x7f)
{
@@ -1552,8 +1547,6 @@ s32 seq_channel_layer_process_script_part3(struct SequenceChannelLayer *layer, s
u8 get_instrument(struct SequenceChannel *seqChannel, u8 instId, struct Instrument **instOut, struct AdsrSettings *adsr)
{
DEBUG_PRINT("@ get_instrument()");
DEBUG_PRINT("- instrument id: %d", instId);
struct Instrument *inst;
#if defined(VERSION_EU) || defined(VERSION_SH)
@@ -1619,9 +1612,6 @@ u8 get_instrument(struct SequenceChannel *seqChannel, u8 instId, struct Instrume
void set_instrument(struct SequenceChannel *seqChannel, u8 instId)
{
DEBUG_PRINT("@ set_instrument()");
DEBUG_PRINT("- bank id: %d", seqChannel->bankId);
DEBUG_PRINT("- instrument id: %d", instId);
if (instId >= 0x80)
{
@@ -1658,7 +1648,6 @@ void sequence_channel_set_volume(struct SequenceChannel *seqChannel, u8 volume)
void sequence_channel_process_script(struct SequenceChannel *seqChannel)
{
DEBUG_PRINT("sequence_channel_process_script()");
s8 temp;
u8 loBits;
@@ -1702,7 +1691,6 @@ void sequence_channel_process_script(struct SequenceChannel *seqChannel)
for (;;)
{
u8 cmd = m64_read_u8(state);
DEBUG_PRINT("- handling command: %x", cmd);
#if !defined(VERSION_EU) && !defined(VERSION_SH)
if (cmd == 0xff) // chan_end
@@ -1895,10 +1883,8 @@ void sequence_channel_process_script(struct SequenceChannel *seqChannel)
#endif
case 0xc1: // chan_setinstr ("set program"?)
DEBUG_PRINT(" - cmd 0xc1, setinstr");
u8 instrId = m64_read_u8(state);
DEBUG_PRINT(" - read %d from state", instrId);
set_instrument(seqChannel, instrId);
break;
@@ -2030,12 +2016,9 @@ void sequence_channel_process_script(struct SequenceChannel *seqChannel)
break;
case 0xc6: // chan_setbank; switch bank within set
DEBUG_PRINT(" - case 0xc6 - switch bank");
DEBUG_PRINT(" - seq id %d", seqPlayer->seqId);
cmd = m64_read_u8(state);
DEBUG_PRINT(" - backwards bank id %d", cmd);
// Switch to the temp's (0-indexed) bank in this sequence's
// bank set. Note that in the binary format (not in the JSON!)
@@ -2362,8 +2345,6 @@ void sequence_channel_process_script(struct SequenceChannel *seqChannel)
#else
case 0x80: // chan_ioreadval; read data from audio lib
#endif
DEBUG_PRINT("- cmd 0x80, read data from audio lib");
DEBUG_PRINT(" - reading index: %d", loBits);
value = seqChannel->soundScriptIO[loBits];
if (loBits < 4)
{
@@ -2463,7 +2444,6 @@ out:
void sequence_player_process_sequence(struct SequencePlayer *seqPlayer)
{
DEBUG_PRINT("sequence_player_process_sequence()");
u8 cmd;
#ifdef VERSION_SH
-50
View File
@@ -28,13 +28,9 @@
#define DMEM_ADDR_WET_RIGHT_CH 0x880
#define aSetLoadBufferPair(pkt, c, off) \
DEBUG_PRINT("- (in set load buffer pair, set buffer 1) "); \
aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_LEFT_CH, 0, DEFAULT_LEN_1CH - c); \
DEBUG_PRINT("- (in set load buffer pair, load buffer 1) "); \
aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.left + (off))); \
DEBUG_PRINT("- (in set load buffer pair, set buffer 2) "); \
aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_RIGHT_CH, 0, DEFAULT_LEN_1CH - c); \
DEBUG_PRINT("- (in set load buffer pair, load buffer 2) "); \
aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.right + (off)))
#define aSetSaveBufferPair(pkt, c, d, off) \
@@ -362,7 +358,6 @@ aiBufPtr+= chunkLen;
// bufLen will be divisible by 16
u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen)
{
DEBUG_PRINT("synthesis_execute()");
s32 chunkLen;
u32 *aiBufPtr = (u32 *)aiBuf;
u64 *cmd = cmdBuf + 1;
@@ -583,66 +578,51 @@ u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateI
#else
u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateIndex)
{
DEBUG_PRINT("synthesis_do_one_audio_update()");
UNUSED s32 pad1[1];
UNUSED s32 pad[2];
UNUSED s32 pad2[1];
s16 temp;
DEBUG_PRINT("- curFrame: %d", gSynthesisReverb.curFrame);
DEBUG_PRINT("- updateIndex: %d", updateIndex);
struct ReverbRingBufferItem *v1 = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
DEBUG_PRINT("- v1: %x", v1);
if (gSynthesisReverb.useReverb == 0)
{
DEBUG_PRINT("- w/o reverb");
aClearBuffer(cmd++, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
}
else
{
DEBUG_PRINT("- w/ reverb");
if (gReverbDownsampleRate == 1)
{
DEBUG_PRINT("- w/ reverb downsample");
// Put the oldest samples in the ring buffer into the wet channels
DEBUG_PRINT("- set load buffer pair 1");
DEBUG_PRINT("- startPos: %d", v1->startPos);
aSetLoadBufferPair(cmd++, 0, v1->startPos);
if (v1->lengthB != 0)
{
// Ring buffer wrapped
DEBUG_PRINT("- set load buffer pair 2");
aSetLoadBufferPair(cmd++, v1->lengthA, 0);
temp = 0;
}
// Use the reverb sound as initial sound for this audio update
DEBUG_PRINT("- dmem move");
aDMEMMove(cmd++, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
// (Hopefully) lower the volume of the wet channels. New reverb will later be mixed into
// these channels.
DEBUG_PRINT("- set buffer");
aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
// 0x8000 here is -100%
DEBUG_PRINT("- mix");
aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_WET_LEFT_CH,
/*out*/ DMEM_ADDR_WET_LEFT_CH);
}
else
{
DEBUG_PRINT("- w/o reverb downsample");
// Same as above but upsample the previously downsampled samples used for reverb first
temp = 0; //! jesus christ
s16 t4 = (v1->startPos & 7) * 2;
s16 ra = ALIGN(v1->lengthA + t4, 4);
DEBUG_PRINT("- set load buffer pair");
aSetLoadBufferPair(cmd++, 0, v1->startPos - t4 / 2);
if (v1->lengthB != 0)
{
@@ -654,19 +634,12 @@ u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateI
//! useless assignment.
ra = ra + temp;
}
DEBUG_PRINT("- set buffer 1");
aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, bufLen << 1);
DEBUG_PRINT("- resample 1");
aResample(cmd++, gSynthesisReverb.resampleFlags, gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateLeft));
DEBUG_PRINT("- set buffer 2");
aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_RIGHT_CH, DMEM_ADDR_RIGHT_CH, bufLen << 1);
DEBUG_PRINT("- resample 2");
aResample(cmd++, gSynthesisReverb.resampleFlags, gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateRight));
DEBUG_PRINT("- set buffer 3");
aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
DEBUG_PRINT("- mix");
aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_LEFT_CH, /*out*/ DMEM_ADDR_LEFT_CH);
DEBUG_PRINT("- dmem move");
aDMEMMove(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
}
cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
@@ -700,7 +673,6 @@ u64 *synthesis_process_note(struct Note *note, struct NoteSubEu *noteSubEu, stru
#else
u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
{
DEBUG_PRINT("synthesis_process_notes()");
s32 noteIndex; // sp174
struct Note *note; // s7
@@ -787,29 +759,22 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
for (noteIndex = 0; noteIndex < gMaxSimultaneousNotes; noteIndex++)
{
DEBUG_PRINT("- for note index %d/%d", noteIndex, gMaxSimultaneousNotes);
DEBUG_PRINT("- getting note");
note = &gNotes[noteIndex];
//! This function requires note->enabled to be volatile, but it breaks other functions like note_enable.
//! Casting to a struct with just the volatile bitfield works, but there may be a better way to match.
DEBUG_PRINT("- if note is enabled but not loaded");
if (((struct vNote *)note)->enabled && IS_BANK_LOAD_COMPLETE(note->bankId) == FALSE)
{
DEBUG_PRINT("- note is enabled but not loaded");
gAudioErrorFlags = (note->bankId << 8) + noteIndex + 0x1000000;
continue;
}
DEBUG_PRINT("- if note is enabled");
if (((struct vNote *)note)->enabled)
{
DEBUG_PRINT("$ note is enabled!");
flags = 0;
DEBUG_PRINT("- if note needs to be init");
if (note->needsInit == TRUE)
{
flags = A_INIT;
@@ -817,7 +782,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
note->samplePosFrac = 0;
}
DEBUG_PRINT("- if note frequency is less than 2");
if (note->frequency < US_FLOAT(2.0))
{
nParts = 1;
@@ -842,12 +806,10 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
samplesLenFixedPoint = note->samplePosFrac + resamplingRateFixedPoint * bufLen * 2;
note->samplePosFrac = samplesLenFixedPoint & 0xFFFF; // 16-bit store, can't reuse
DEBUG_PRINT("- if note sound is null");
if (note->sound == NULL)
{
// A wave synthesis note (not ADPCM)
DEBUG_PRINT("- note is null, do wave synthesis");
cmd = load_wave_samples(cmd, note, samplesLenFixedPoint >> 0x10);
noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_UNCOMPRESSED_NOTE + note->samplePosInt * 2;
note->samplePosInt += samplesLenFixedPoint >> 0x10;
@@ -856,7 +818,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
else
{
// ADPCM note
DEBUG_PRINT("- @ handle adpcm note");
audioBookSample = note->sound->sample;
@@ -866,7 +827,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
resampledTempLen = 0;
for (curPart = 0; curPart < nParts; curPart++)
{
DEBUG_PRINT("- for part %d", curPart);
nAdpcmSamplesProcessed = 0; // s8
s5 = 0; // s4
@@ -888,7 +848,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
u32 nEntries; // v1
curLoadedBook = audioBookSample->book->book;
nEntries = audioBookSample->book->order * audioBookSample->book->npredictors;
DEBUG_PRINT("- loading adpcm");
aLoadADPCM(cmd++, nEntries * 16, VIRTUAL_TO_PHYSICAL2(curLoadedBook));
}
@@ -952,7 +911,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
a3 = 0;
}
DEBUG_PRINT("- if not note restart");
if (note->restart != FALSE)
{
aSetLoop(cmd++, VIRTUAL_TO_PHYSICAL2(audioBookSample->loop->state));
@@ -1000,7 +958,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
}
flags = 0;
DEBUG_PRINT("- if note finished");
if (noteFinished)
{
aClearBuffer(cmd++, DMEM_ADDR_UNCOMPRESSED_NOTE + s5,
@@ -1011,7 +968,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
break;
}
DEBUG_PRINT("- if restart");
if (restart)
{
note->restart = TRUE;
@@ -1097,20 +1053,14 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
}
}
DEBUG_PRINT("- done handling notes");
DEBUG_PRINT("- setting buffer 1");
t9 = bufLen * 2;
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, t9);
DEBUG_PRINT("- interleaving");
aInterleave(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_RIGHT_CH);
t9 *= 2;
DEBUG_PRINT("- setting buffer 2");
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, t9);
DEBUG_PRINT("- saving buffer");
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(aiBuf));
DEBUG_PRINT("- returning from process notes");
return cmd;
}