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DSP: Deduplicate the accelerator code

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The logic is entirely the same; only the inputs and outputs are
different, so deduplicating makes sense.

This will make fixing accelerator issues easier.
This commit is contained in:
Léo Lam 2017-09-16 16:40:48 +02:00
commit 38a7196ec6
3 changed files with 77 additions and 140 deletions
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89
Source/Core/Core/HW/DSPHLE/UCodes/AXVoice.h
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@ -16,6 +16,7 @@
#include "Common/CommonTypes.h"
#include "Common/MathUtil.h"
#include "Core/DSP/DSPAccelerator.h"
#include "Core/HW/DSP.h"
#include "Core/HW/DSPHLE/UCodes/AX.h"
#include "Core/HW/DSPHLE/UCodes/AXStructs.h"
@ -180,98 +181,22 @@ void AcceleratorSetup(PB_TYPE* pb, u32* cur_addr)
acc_end_reached = false;
}
// Reads a sample from the simulated accelerator. Also handles looping and
// Reads a sample from the accelerator. Also handles looping and
// disabling streams that reached the end (this is done by an exception raised
// by the accelerator on real hardware).
u16 AcceleratorGetSample()
{
u16 ret;
u8 step_size_bytes = 0;
// See below for explanations about acc_end_reached.
if (acc_end_reached)
return 0;
switch (acc_pb->audio_addr.sample_format)
{
case 0x00: // ADPCM
{
// ADPCM decoding, not much to explain here.
if ((*acc_cur_addr & 15) == 0)
{
acc_pb->adpcm.pred_scale = DSP::ReadARAM((*acc_cur_addr & ~15) >> 1);
*acc_cur_addr += 2;
}
switch (acc_end_addr & 15)
{
case 0: // Tom and Jerry
step_size_bytes = 1;
break;
case 1: // Blazing Angels
step_size_bytes = 0;
break;
default:
step_size_bytes = 2;
break;
}
int scale = 1 << (acc_pb->adpcm.pred_scale & 0xF);
int coef_idx = (acc_pb->adpcm.pred_scale >> 4) & 0x7;
s32 coef1 = acc_pb->adpcm.coefs[coef_idx * 2 + 0];
s32 coef2 = acc_pb->adpcm.coefs[coef_idx * 2 + 1];
int temp = (*acc_cur_addr & 1) ? (DSP::ReadARAM(*acc_cur_addr >> 1) & 0xF) :
(DSP::ReadARAM(*acc_cur_addr >> 1) >> 4);
if (temp >= 8)
temp -= 16;
int val =
(scale * temp) + ((0x400 + coef1 * acc_pb->adpcm.yn1 + coef2 * acc_pb->adpcm.yn2) >> 11);
val = MathUtil::Clamp(val, -0x7FFF, 0x7FFF);
acc_pb->adpcm.yn2 = acc_pb->adpcm.yn1;
acc_pb->adpcm.yn1 = val;
*acc_cur_addr += 1;
ret = val;
break;
}
case 0x0A: // 16-bit PCM audio
ret = (DSP::ReadARAM(*acc_cur_addr * 2) << 8) | DSP::ReadARAM(*acc_cur_addr * 2 + 1);
acc_pb->adpcm.yn2 = acc_pb->adpcm.yn1;
acc_pb->adpcm.yn1 = ret;
step_size_bytes = 2;
*acc_cur_addr += 1;
break;
case 0x19: // 8-bit PCM audio
ret = DSP::ReadARAM(*acc_cur_addr) << 8;
acc_pb->adpcm.yn2 = acc_pb->adpcm.yn1;
acc_pb->adpcm.yn1 = ret;
step_size_bytes = 2;
*acc_cur_addr += 1;
break;
default:
ERROR_LOG(DSPHLE, "Unknown sample format: %d", acc_pb->audio_addr.sample_format);
return 0;
}
// Have we reached the end address?
//
// On real hardware, this would raise an interrupt that is handled by the
// UCode. We simulate what this interrupt does here.
if (*acc_cur_addr == (acc_end_addr + step_size_bytes - 1))
{
auto end_address_reached = [] {
// loop back to loop_addr.
*acc_cur_addr = acc_loop_addr;
if (acc_pb->audio_addr.looping)
{
// Set the ADPCM infos to continue processing at loop_addr.
// Set the ADPCM info to continue processing at loop_addr.
//
// For some reason, yn1 and yn2 aren't set if the voice is not of
// stream type. This is what the AX UCode does and I don't really
@ -304,9 +229,11 @@ u16 AcceleratorGetSample()
acc_end_reached = true;
#endif
}
}
};
return ret;
return ReadAccelerator(acc_loop_addr, acc_end_addr, acc_cur_addr,
acc_pb->audio_addr.sample_format, &acc_pb->adpcm.yn1, &acc_pb->adpcm.yn2,
&acc_pb->adpcm.pred_scale, acc_pb->adpcm.coefs, end_address_reached);
}
// Reads samples from the input callback, resamples them to <count> samples at