// 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(); } } }