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@@ -28,13 +28,9 @@
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#define DMEM_ADDR_WET_RIGHT_CH 0x880
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#define aSetLoadBufferPair(pkt, c, off) \
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DEBUG_PRINT("- (in set load buffer pair, set buffer 1) "); \
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aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_LEFT_CH, 0, DEFAULT_LEN_1CH - c); \
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DEBUG_PRINT("- (in set load buffer pair, load buffer 1) "); \
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aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.left + (off))); \
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DEBUG_PRINT("- (in set load buffer pair, set buffer 2) "); \
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aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_RIGHT_CH, 0, DEFAULT_LEN_1CH - c); \
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DEBUG_PRINT("- (in set load buffer pair, load buffer 2) "); \
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aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.right + (off)))
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#define aSetSaveBufferPair(pkt, c, d, off) \
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@@ -362,7 +358,6 @@ aiBufPtr+= chunkLen;
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// bufLen will be divisible by 16
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u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen)
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{
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DEBUG_PRINT("synthesis_execute()");
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s32 chunkLen;
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u32 *aiBufPtr = (u32 *)aiBuf;
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u64 *cmd = cmdBuf + 1;
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@@ -583,66 +578,51 @@ u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateI
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#else
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u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateIndex)
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{
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DEBUG_PRINT("synthesis_do_one_audio_update()");
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UNUSED s32 pad1[1];
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UNUSED s32 pad[2];
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UNUSED s32 pad2[1];
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s16 temp;
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DEBUG_PRINT("- curFrame: %d", gSynthesisReverb.curFrame);
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DEBUG_PRINT("- updateIndex: %d", updateIndex);
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struct ReverbRingBufferItem *v1 = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
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DEBUG_PRINT("- v1: %x", v1);
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if (gSynthesisReverb.useReverb == 0)
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{
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DEBUG_PRINT("- w/o reverb");
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aClearBuffer(cmd++, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
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cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
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}
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else
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{
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DEBUG_PRINT("- w/ reverb");
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if (gReverbDownsampleRate == 1)
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{
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DEBUG_PRINT("- w/ reverb downsample");
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// Put the oldest samples in the ring buffer into the wet channels
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DEBUG_PRINT("- set load buffer pair 1");
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DEBUG_PRINT("- startPos: %d", v1->startPos);
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aSetLoadBufferPair(cmd++, 0, v1->startPos);
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if (v1->lengthB != 0)
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{
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// Ring buffer wrapped
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DEBUG_PRINT("- set load buffer pair 2");
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aSetLoadBufferPair(cmd++, v1->lengthA, 0);
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temp = 0;
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}
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// Use the reverb sound as initial sound for this audio update
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DEBUG_PRINT("- dmem move");
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aDMEMMove(cmd++, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
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// (Hopefully) lower the volume of the wet channels. New reverb will later be mixed into
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// these channels.
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DEBUG_PRINT("- set buffer");
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aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
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// 0x8000 here is -100%
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DEBUG_PRINT("- mix");
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aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_WET_LEFT_CH,
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/*out*/ DMEM_ADDR_WET_LEFT_CH);
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}
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else
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{
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DEBUG_PRINT("- w/o reverb downsample");
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// Same as above but upsample the previously downsampled samples used for reverb first
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temp = 0; //! jesus christ
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s16 t4 = (v1->startPos & 7) * 2;
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s16 ra = ALIGN(v1->lengthA + t4, 4);
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DEBUG_PRINT("- set load buffer pair");
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aSetLoadBufferPair(cmd++, 0, v1->startPos - t4 / 2);
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if (v1->lengthB != 0)
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{
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@@ -654,19 +634,12 @@ u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateI
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//! useless assignment.
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ra = ra + temp;
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}
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DEBUG_PRINT("- set buffer 1");
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aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, bufLen << 1);
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DEBUG_PRINT("- resample 1");
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aResample(cmd++, gSynthesisReverb.resampleFlags, gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateLeft));
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DEBUG_PRINT("- set buffer 2");
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aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_RIGHT_CH, DMEM_ADDR_RIGHT_CH, bufLen << 1);
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DEBUG_PRINT("- resample 2");
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aResample(cmd++, gSynthesisReverb.resampleFlags, gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateRight));
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DEBUG_PRINT("- set buffer 3");
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aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
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DEBUG_PRINT("- mix");
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aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_LEFT_CH, /*out*/ DMEM_ADDR_LEFT_CH);
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DEBUG_PRINT("- dmem move");
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aDMEMMove(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
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}
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cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
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@@ -700,7 +673,6 @@ u64 *synthesis_process_note(struct Note *note, struct NoteSubEu *noteSubEu, stru
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#else
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u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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{
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DEBUG_PRINT("synthesis_process_notes()");
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s32 noteIndex; // sp174
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struct Note *note; // s7
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@@ -787,29 +759,22 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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for (noteIndex = 0; noteIndex < gMaxSimultaneousNotes; noteIndex++)
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{
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DEBUG_PRINT("- for note index %d/%d", noteIndex, gMaxSimultaneousNotes);
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DEBUG_PRINT("- getting note");
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note = &gNotes[noteIndex];
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//! This function requires note->enabled to be volatile, but it breaks other functions like note_enable.
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//! Casting to a struct with just the volatile bitfield works, but there may be a better way to match.
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DEBUG_PRINT("- if note is enabled but not loaded");
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if (((struct vNote *)note)->enabled && IS_BANK_LOAD_COMPLETE(note->bankId) == FALSE)
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{
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DEBUG_PRINT("- note is enabled but not loaded");
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gAudioErrorFlags = (note->bankId << 8) + noteIndex + 0x1000000;
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continue;
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}
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DEBUG_PRINT("- if note is enabled");
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if (((struct vNote *)note)->enabled)
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{
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DEBUG_PRINT("$ note is enabled!");
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flags = 0;
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DEBUG_PRINT("- if note needs to be init");
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if (note->needsInit == TRUE)
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{
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flags = A_INIT;
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@@ -817,7 +782,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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note->samplePosFrac = 0;
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}
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DEBUG_PRINT("- if note frequency is less than 2");
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if (note->frequency < US_FLOAT(2.0))
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{
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nParts = 1;
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@@ -842,12 +806,10 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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samplesLenFixedPoint = note->samplePosFrac + resamplingRateFixedPoint * bufLen * 2;
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note->samplePosFrac = samplesLenFixedPoint & 0xFFFF; // 16-bit store, can't reuse
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DEBUG_PRINT("- if note sound is null");
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if (note->sound == NULL)
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{
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// A wave synthesis note (not ADPCM)
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DEBUG_PRINT("- note is null, do wave synthesis");
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cmd = load_wave_samples(cmd, note, samplesLenFixedPoint >> 0x10);
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noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_UNCOMPRESSED_NOTE + note->samplePosInt * 2;
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note->samplePosInt += samplesLenFixedPoint >> 0x10;
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@@ -856,7 +818,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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else
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{
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// ADPCM note
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DEBUG_PRINT("- @ handle adpcm note");
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audioBookSample = note->sound->sample;
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@@ -866,7 +827,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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resampledTempLen = 0;
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for (curPart = 0; curPart < nParts; curPart++)
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{
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DEBUG_PRINT("- for part %d", curPart);
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nAdpcmSamplesProcessed = 0; // s8
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s5 = 0; // s4
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@@ -888,7 +848,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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u32 nEntries; // v1
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curLoadedBook = audioBookSample->book->book;
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nEntries = audioBookSample->book->order * audioBookSample->book->npredictors;
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DEBUG_PRINT("- loading adpcm");
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aLoadADPCM(cmd++, nEntries * 16, VIRTUAL_TO_PHYSICAL2(curLoadedBook));
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}
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@@ -952,7 +911,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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a3 = 0;
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}
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DEBUG_PRINT("- if not note restart");
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if (note->restart != FALSE)
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{
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aSetLoop(cmd++, VIRTUAL_TO_PHYSICAL2(audioBookSample->loop->state));
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@@ -1000,7 +958,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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}
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flags = 0;
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DEBUG_PRINT("- if note finished");
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if (noteFinished)
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{
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aClearBuffer(cmd++, DMEM_ADDR_UNCOMPRESSED_NOTE + s5,
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@@ -1011,7 +968,6 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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break;
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}
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DEBUG_PRINT("- if restart");
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if (restart)
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{
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note->restart = TRUE;
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@@ -1097,20 +1053,14 @@ u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
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}
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}
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DEBUG_PRINT("- done handling notes");
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DEBUG_PRINT("- setting buffer 1");
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t9 = bufLen * 2;
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aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, t9);
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DEBUG_PRINT("- interleaving");
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aInterleave(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_RIGHT_CH);
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t9 *= 2;
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DEBUG_PRINT("- setting buffer 2");
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aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, t9);
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DEBUG_PRINT("- saving buffer");
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aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(aiBuf));
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DEBUG_PRINT("- returning from process notes");
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return cmd;
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}
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