// license:BSD-3-Clause
// copyright-holders:Olivier Galibert
/*
GI SP0250 digital LPC sound synthesizer
By O. Galibert.
Unknown:
- Exact clock divider
- Exact noise algorithm
- Exact noise pitch (probably ok)
- 7 bits output mapping
- Whether the pitch starts counting from 0 or 1
Unimplemented:
- Direct Data test mode (pin 7)
Sound quite reasonably already though.
*/
#include "emu.h"
#include "sp0250.h"
/*
standard external clock is 3.12MHz
the chip provides a 445.7kHz output clock, which is = 3.12MHz / 7
therefore I expect the clock divider to be a multiple of 7
Also there are 6 cascading filter stages so I expect the divider to be a multiple of 6.
The SP0250 manual states that the original speech is sampled at 10kHz, so the divider
should be 312, but 312 = 39*8 so it doesn't look right because a divider by 39 is unlikely.
7*6*8 = 336 gives a 9.286kHz sample rate and matches the samples from the Sega boards.
*/
#define CLOCK_DIVIDER (7*6*8)
DEFINE_DEVICE_TYPE(SP0250, sp0250_device, "sp0250", "GI SP0250 LPC")
sp0250_device::sp0250_device(const machine_config &mconfig, const char *tag, device_t *owner, uint32_t clock) :
device_t(mconfig, SP0250, tag, owner, clock),
device_sound_interface(mconfig, *this),
m_amp(0),
m_pitch(0),
m_repeat(0),
m_pcount(0),
m_rcount(0),
m_playing(0),
m_RNG(0),
m_stream(nullptr),
m_voiced(0),
m_fifo_pos(0),
m_drq(*this)
{
for (auto & elem : m_fifo)
{
elem = 0;
}
for (auto & elem : m_filter)
{
elem.F = 0;
elem.B = 0;
elem.z1 = 0;
elem.z2 = 0;
}
}
//-------------------------------------------------
// device_start - device-specific startup
//-------------------------------------------------
void sp0250_device::device_start()
{
m_RNG = 1;
m_drq.resolve_safe();
if (!m_drq.isnull())
{
m_drq(ASSERT_LINE);
m_tick_timer= machine().scheduler().timer_alloc(timer_expired_delegate(FUNC(sp0250_device::timer_tick), this));
m_tick_timer->adjust(attotime::from_hz(clock()) * CLOCK_DIVIDER, 0, attotime::from_hz(clock()) * CLOCK_DIVIDER);
}
m_stream = machine().sound().stream_alloc(*this, 0, 1, clock() / CLOCK_DIVIDER);
save_item(NAME(m_amp));
save_item(NAME(m_pitch));
save_item(NAME(m_repeat));
save_item(NAME(m_pcount));
save_item(NAME(m_rcount));
save_item(NAME(m_playing));
save_item(NAME(m_RNG));
save_item(NAME(m_voiced));
save_item(NAME(m_fifo));
save_item(NAME(m_fifo_pos));
}
static uint16_t sp0250_ga(uint8_t v)
{
return (v & 0x1f) << (v>>5);
}
static int16_t sp0250_gc(uint8_t v)
{
// Internal ROM to the chip, cf. manual
static const uint16_t coefs[128] =
{
0, 9, 17, 25, 33, 41, 49, 57, 65, 73, 81, 89, 97, 105, 113, 121,
129, 137, 145, 153, 161, 169, 177, 185, 193, 201, 203, 217, 225, 233, 241, 249,
257, 265, 273, 281, 289, 297, 301, 305, 309, 313, 317, 321, 325, 329, 333, 337,
341, 345, 349, 353, 357, 361, 365, 369, 373, 377, 381, 385, 389, 393, 397, 401,
405, 409, 413, 417, 421, 425, 427, 429, 431, 433, 435, 437, 439, 441, 443, 445,
447, 449, 451, 453, 455, 457, 459, 461, 463, 465, 467, 469, 471, 473, 475, 477,
479, 481, 482, 483, 484, 485, 486, 487, 488, 489, 490, 491, 492, 493, 494, 495,
496, 497, 498, 499, 500, 501, 502, 503, 504, 505, 506, 507, 508, 509, 510, 511
};
int16_t res = coefs[v & 0x7f];
if (!(v & 0x80))
res = -res;
return res;
}
void sp0250_device::load_values()
{
m_filter[0].B = sp0250_gc(m_fifo[ 0]);
m_filter[0].F = sp0250_gc(m_fifo[ 1]);
m_amp = sp0250_ga(m_fifo[ 2]);
m_filter[1].B = sp0250_gc(m_fifo[ 3]);
m_filter[1].F = sp0250_gc(m_fifo[ 4]);
m_pitch = m_fifo[ 5];
m_filter[2].B = sp0250_gc(m_fifo[ 6]);
m_filter[2].F = sp0250_gc(m_fifo[ 7]);
m_repeat = m_fifo[ 8] & 0x3f;
m_voiced = m_fifo[ 8] & 0x40;
m_filter[3].B = sp0250_gc(m_fifo[ 9]);
m_filter[3].F = sp0250_gc(m_fifo[10]);
m_filter[4].B = sp0250_gc(m_fifo[11]);
m_filter[4].F = sp0250_gc(m_fifo[12]);
m_filter[5].B = sp0250_gc(m_fifo[13]);
m_filter[5].F = sp0250_gc(m_fifo[14]);
m_fifo_pos = 0;
m_drq(ASSERT_LINE);
m_pcount = 0;
m_rcount = 0;
for (int f = 0; f < 6; f++)
m_filter[f].z1 = m_filter[f].z2 = 0;
m_playing = 1;
}
TIMER_CALLBACK_MEMBER( sp0250_device::timer_tick )
{
m_stream->update();
}
WRITE8_MEMBER( sp0250_device::write )
{
m_stream->update();
if (m_fifo_pos != 15)
{
m_fifo[m_fifo_pos++] = data;
if (m_fifo_pos == 15)
m_drq(CLEAR_LINE);
}
else
logerror("%s: overflow SP0250 FIFO\n", machine().describe_context());
}
uint8_t sp0250_device::drq_r()
{
m_stream->update();
return (m_fifo_pos == 15) ? CLEAR_LINE : ASSERT_LINE;
}
//-------------------------------------------------
// sound_stream_update - handle a stream update
//-------------------------------------------------
void sp0250_device::sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples)
{
stream_sample_t *output = outputs[0];
for (int i = 0; i < samples; i++)
{
if (m_playing)
{
int16_t z0;
if (m_voiced)
{
if(!m_pcount)
z0 = m_amp;
else
z0 = 0;
}
else
{
// Borrowing the ay noise generation LFSR
if(m_RNG & 1)
{
z0 = m_amp;
m_RNG ^= 0x24000;
}
else
z0 = -m_amp;
m_RNG >>= 1;
}
for (int f = 0; f < 6; f++)
{
z0 += ((m_filter[f].z1 * m_filter[f].F) >> 8)
+ ((m_filter[f].z2 * m_filter[f].B) >> 9);
m_filter[f].z2 = m_filter[f].z1;
m_filter[f].z1 = z0;
}
// Physical resolution is only 7 bits, but heh
// max amplitude is 0x0f80 so we have margin to push up the output
output[i] = z0 << 3;
m_pcount++;
if (m_pcount >= m_pitch)
{
m_pcount = 0;
m_rcount++;
if (m_rcount >= m_repeat)
m_playing = 0;
}
}
else
output[i] = 0;
if (!m_playing)
{
if(m_fifo_pos == 15)
load_values();
}
}
}