ShuriZma
/
dolphin
mirror of https://github.com/dolphin-emu/dolphin.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity

Merge pull request #985 from skidau/dsp-loop-hle 

Moved the loop check back to a post-loop condition in DSP HLE AX ucode.
This commit is contained in:
shuffle2 2014-09-06 11:44:27 -07:00
parent 87b6c8d991 9800252a28
commit f2237ecd64
1 changed files with 60 additions and 61 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

121
Source/Core/Core/HW/DSPHLE/UCodes/AXVoice.h
View File

@ -149,11 +149,67 @@ void AcceleratorSetup(PB_TYPE* pb, u32* cur_addr)
u16 AcceleratorGetSample()
{
u16 ret;
u8 step_size_bytes = 2;
u8 step_size_bytes = 0;
// 8-bit PCM audio uses 1 byte per sample/sample block, not 2 like other formats.
if (acc_pb->audio_addr.sample_format == 0x19)
step_size_bytes = 1;
// 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;
}
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);
MathUtil::Clamp(&val, -0x7FFF, 0x7FFF);
acc_pb->adpcm.yn2 = acc_pb->adpcm.yn1;
acc_pb->adpcm.yn1 = val;
step_size_bytes = 2;
*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 = 1;
*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?
//
@ -194,63 +250,6 @@ u16 AcceleratorGetSample()
}
}
// See above 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;
}
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);
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;
*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;
*acc_cur_addr += 1;
break;
default:
ERROR_LOG(DSPHLE, "Unknown sample format: %d", acc_pb->audio_addr.sample_format);
return 0;
}
return ret;
}