From 7ce27dadde9c2ad7f8b75e963bae8f47638d054d Mon Sep 17 00:00:00 2001 From: Devin Acker Date: Tue, 14 Dec 2021 10:25:57 -0500 Subject: [PATCH] ctk551: implement sound, promote to working (#8960) --- src/devices/cpu/h8/gt913.cpp | 19 +- src/devices/cpu/h8/gt913.h | 2 +- src/devices/cpu/h8/gt913.lst | 2 +- src/devices/machine/gt913_io.cpp | 4 +- src/devices/machine/gt913_snd.cpp | 323 +++++++++++++++++++++++------- src/devices/machine/gt913_snd.h | 69 +++++-- src/mame/drivers/ctk551.cpp | 20 +- 7 files changed, 340 insertions(+), 99 deletions(-) diff --git a/src/devices/cpu/h8/gt913.cpp b/src/devices/cpu/h8/gt913.cpp index bc8e11f5dc5..bc577206e48 100644 --- a/src/devices/cpu/h8/gt913.cpp +++ b/src/devices/cpu/h8/gt913.cpp @@ -11,7 +11,7 @@ - Two timers, three 8-bit ports, two 8-bit ADCs - Keyboard controller w/ key velocity detection - MIDI UART - - 24-voice PCM sound (currently not emulated / fully understood) + - 24-voice DPCM sound Earlier and later Casio keyboard models contain "uPD912" and "uPD914" chips, which are presumably similar. @@ -52,9 +52,9 @@ void gt913_device::map(address_map &map) map(0xfac0, 0xffbf).ram(); // CTK-551 zeroes out this range at $0418 /* ffc0-ffcb: sound */ - map(0xffc0, 0xffc5).rw(m_sound, FUNC(gt913_sound_hle_device::data_r), FUNC(gt913_sound_hle_device::data_w)); - map(0xffc6, 0xffc7).w(m_sound, FUNC(gt913_sound_hle_device::command_w)); - map(0xffca, 0xffcb).r(m_sound, FUNC(gt913_sound_hle_device::status_r)); + map(0xffc0, 0xffc5).rw(m_sound, FUNC(gt913_sound_device::data_r), FUNC(gt913_sound_device::data_w)); + map(0xffc6, 0xffc7).w(m_sound, FUNC(gt913_sound_device::command_w)); + map(0xffca, 0xffcb).r(m_sound, FUNC(gt913_sound_device::status_r)); /* ffd0-ffd5: key controller */ map(0xffd0, 0xffd1).r(m_kbd, FUNC(gt913_kbd_hle_device::read)); @@ -88,7 +88,14 @@ void gt913_device::device_add_mconfig(machine_config &config) { GT913_INTC(config, "intc"); - GT913_SOUND_HLE(config, m_sound, 0); + /* + generate sound at 104 cycles per sample (= 144.231 kHz sample clock to the DAC) + on keyboard models that include a DSP, this also results in a multiple + of the 36.058 kHz CPU->DSP sync signal shown in some schematics (WK-1200 and others) + */ + GT913_SOUND(config, m_sound, std::round(clock() / 104.0f)); + m_sound->set_device_rom_tag(tag()); + GT913_KBD_HLE(config, m_kbd, 0); m_kbd->irq_cb().set([this](int val) { if (val) m_intc->internal_interrupt(5); }); GT913_IO_HLE(config, m_io_hle, "intc", 6, 7); @@ -102,7 +109,7 @@ void gt913_device::device_add_mconfig(machine_config &config) void gt913_device::uart_rate_w(uint8_t data) { - m_sci->brr_w(data >> 1); + m_sci->brr_w(data >> 2); } void gt913_device::uart_control_w(uint8_t data) diff --git a/src/devices/cpu/h8/gt913.h b/src/devices/cpu/h8/gt913.h index 3269ab7eafe..ffb7e1b3f6b 100644 --- a/src/devices/cpu/h8/gt913.h +++ b/src/devices/cpu/h8/gt913.h @@ -62,7 +62,7 @@ protected: required_device m_intc; /* sound */ - required_device m_sound; + required_device m_sound; /* key controller */ required_device m_kbd; diff --git a/src/devices/cpu/h8/gt913.lst b/src/devices/cpu/h8/gt913.lst index 7bbb26ecc33..45f1ddc995c 100644 --- a/src/devices/cpu/h8/gt913.lst +++ b/src/devices/cpu/h8/gt913.lst @@ -107,7 +107,7 @@ 5e00 ff00 0 bsr rel8 - 5f00 fff0 0 mov.w imm16 r16l -66000000 ff00ff8f 0 btst imm3 abs8 +66000000 ff00ff8f 0 btst imm3 r16ihh 6800 ff80 0 mov.b r16ih r8l 6900 ff88 0 mov.w r16ih r16l diff --git a/src/devices/machine/gt913_io.cpp b/src/devices/machine/gt913_io.cpp index 211d17de1ac..8c1cef4e556 100644 --- a/src/devices/machine/gt913_io.cpp +++ b/src/devices/machine/gt913_io.cpp @@ -108,9 +108,9 @@ void gt913_io_hle_device::timer_adjust(offs_t num) logerror("unknown timer %u prescaler %u (pc = %04x)\n", num, m_timer_control[num] & 0x7, m_cpu->pc()); [[fallthrough]]; case 0: - clocks <<= 1; break; + break; case 2: - clocks <<= 10; break; + clocks <<= 9; break; } attotime period = m_cpu->clocks_to_attotime(clocks); diff --git a/src/devices/machine/gt913_snd.cpp b/src/devices/machine/gt913_snd.cpp index e8b298da3e5..f06930d3697 100644 --- a/src/devices/machine/gt913_snd.cpp +++ b/src/devices/machine/gt913_snd.cpp @@ -8,13 +8,13 @@ which is then input to either a serial DAC or a HG51B-based DSP, depending on the model of keyboard. - Currently, the actual sample format in ROM is unknown. - The serial output is twos-complement 16-bit PCM, but the data in ROM - doesn't seem to be - reading it as such produces sounds that are - somewhat recognizable, but highly distorted. + The sample format, as well as other details such as the linear interpolation, + are covered in these two Japanese patents: + https://patents.google.com/patent/JP3603343B2/en + https://patents.google.com/patent/JPH07199996A/en - For now, all known (and unknown) register writes are just logged - without generating any sound. + TODO: Volume envelope rates still need adjusting. + (See comment in gt913_sound_device::command_w regarding command 6007) ***************************************************************************/ @@ -26,140 +26,319 @@ // DEVICE DEFINITIONS //************************************************************************** -DEFINE_DEVICE_TYPE(GT913_SOUND_HLE, gt913_sound_hle_device, "gt913_sound_hle", "Casio GT913F sound (HLE)") +DEFINE_DEVICE_TYPE(GT913_SOUND, gt913_sound_device, "gt913_sound_hle", "Casio GT913F sound") -gt913_sound_hle_device::gt913_sound_hle_device(const machine_config &mconfig, const char *tag, device_t *owner, uint32_t clock) : - device_t(mconfig, GT913_SOUND_HLE, tag, owner, clock) +// expand 2-bit exponent deltas +const u8 gt913_sound_device::exp_2_to_3[4] = { 0, 1, 2, 7 }; + +// sign-extend 7-bit sample deltas +const s8 gt913_sound_device::sample_7_to_8[128] = +{ + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, + 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, + 32, 33, 34, 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, + 48, 49, 50, 51, 52, 53, 54, 55, 56, 57, 58, 59, 60, 61, 62, 63, + -64, -63, -62, -61, -60, -59, -58, -57, -56, -55, -54, -53, -52, -51, -50, -49, + -48, -47, -46, -45, -44, -43, -42, -41, -40, -39, -38, -37, -36, -35, -34, -33, + -32, -31, -30, -29, -28, -27, -26, -25, -24, -23, -22, -21, -20, -19, -18, -17, + -16, -15, -14, -13, -12, -11, -10, -9, -8, -7, -6, -5, -4, -3, -2, -1 +}; + +gt913_sound_device::gt913_sound_device(const machine_config &mconfig, const char *tag, device_t *owner, u32 clock) + : device_t(mconfig, GT913_SOUND, tag, owner, clock) + , device_sound_interface(mconfig, *this) + , device_rom_interface(mconfig, *this) { } -void gt913_sound_hle_device::device_start() +void gt913_sound_device::device_start() { + m_stream = stream_alloc(0, 2, clock()); + save_item(NAME(m_gain)); save_item(NAME(m_data)); - save_item(NAME(m_volume_target)); - save_item(NAME(m_volume_rate)); + save_item(STRUCT_MEMBER(m_voices, m_enable)); + + save_item(STRUCT_MEMBER(m_voices, m_addr_start)); + save_item(STRUCT_MEMBER(m_voices, m_addr_end)); + save_item(STRUCT_MEMBER(m_voices, m_addr_loop)); + + save_item(STRUCT_MEMBER(m_voices, m_addr_current)); + save_item(STRUCT_MEMBER(m_voices, m_addr_frac)); + save_item(STRUCT_MEMBER(m_voices, m_pitch)); + + save_item(STRUCT_MEMBER(m_voices, m_sample)); + save_item(STRUCT_MEMBER(m_voices, m_sample_next)); + save_item(STRUCT_MEMBER(m_voices, m_exp)); + + save_item(STRUCT_MEMBER(m_voices, m_volume_current)); + save_item(STRUCT_MEMBER(m_voices, m_volume_target)); + save_item(STRUCT_MEMBER(m_voices, m_volume_rate)); + save_item(STRUCT_MEMBER(m_voices, m_volume_end)); + + save_item(STRUCT_MEMBER(m_voices, m_balance)); + save_item(STRUCT_MEMBER(m_voices, m_gain)); } -void gt913_sound_hle_device::device_reset() +void gt913_sound_device::device_reset() { m_gain = 0; std::memset(m_data, 0, sizeof(m_data)); - std::memset(m_volume_target, 0, sizeof(m_volume_target)); - std::memset(m_volume_rate, 0, sizeof(m_volume_rate)); + + std::memset(m_voices, 0, sizeof(m_voices)); } -void gt913_sound_hle_device::data_w(offs_t offset, uint16_t data) +void gt913_sound_device::sound_stream_update(sound_stream& stream, std::vector const& inputs, std::vector& outputs) +{ + for (int i = 0; i < outputs[0].samples(); i++) + { + s64 left = 0, right = 0; + + for (auto& voice : m_voices) + { + if (voice.m_enable) + mix_sample(voice, left, right); + } + + outputs[0].put_int_clamp(i, (left * m_gain) >> 26, 32678); + outputs[1].put_int_clamp(i, (right * m_gain) >> 26, 32768); + } +} + +void gt913_sound_device::rom_bank_updated() +{ + m_stream->update(); +} + +void gt913_sound_device::mix_sample(voice_t& voice, s64& left, s64& right) +{ + // update sample position + voice.m_addr_frac += voice.m_pitch; + while (voice.m_addr_frac >= (1 << 25)) + { + voice.m_addr_frac -= (1 << 25); + update_sample(voice); + } + + // update volume envelope + if (voice.m_volume_target > voice.m_volume_current + && (voice.m_volume_target - voice.m_volume_current) > voice.m_volume_rate) + { + voice.m_volume_current += voice.m_volume_rate; + } + else if (voice.m_volume_target < voice.m_volume_current + && (voice.m_volume_current - voice.m_volume_target) > voice.m_volume_rate) + { + voice.m_volume_current -= voice.m_volume_rate; + } + else + { + voice.m_volume_current = voice.m_volume_target; + } + + // interpolate, apply envelope + channel gain, and mix into output + const u8 step = (voice.m_addr_frac >> 22) & 7; + const u8 env = (voice.m_volume_current >> 24); + /* + the current envelope level effects amplitude non-linearly, just apply the value twice + (this hardware family is branded as "AČ (A-Square) Sound Source" in some of Casio's + promotional materials, possibly for this reason?) + */ + const s64 sample = ((s64)voice.m_sample + (voice.m_sample_next * step / 8)) * voice.m_gain * env * env; + + left += sample * voice.m_balance[0]; + right += sample * voice.m_balance[1]; +} + +void gt913_sound_device::update_sample(voice_t& voice) +{ + voice.m_sample += voice.m_sample_next; + + if (voice.m_addr_current == (voice.m_addr_loop | 1)) + { + /* + The last 12 bytes of each sample are a table containing five sample and exponent value pairs + for the data words immediately after the loop point. The first pair corresponds to what the + sample and exponent value will be _after_ processing the first word after the loop, + so once we've reached that point, use those values to reload the current sample and exponent + */ + const u32 addr_loop_data = (voice.m_addr_end + 1) & ~1; + + voice.m_sample_next = read_word(addr_loop_data) - voice.m_sample; + voice.m_exp = read_word(addr_loop_data + 10) & 7; + } + else + { + /* + For all other samples, just get the next sample delta value. + For even-numbered samples, also update the exponent/shift value. + */ + const u16 word = read_word(voice.m_addr_current & ~1); + s16 delta = 0; + + if (!BIT(voice.m_addr_current, 0)) + { + voice.m_exp += exp_2_to_3[word & 3]; + voice.m_exp &= 7; + delta = sample_7_to_8[(word >> 2) & 0x7f]; + } + else + { + delta = sample_7_to_8[word >> 9]; + } + + voice.m_sample_next = delta * (1 << voice.m_exp); + } + + voice.m_addr_current++; + if (voice.m_addr_current == voice.m_addr_end) + { + voice.m_addr_current = voice.m_addr_loop; + + if (voice.m_addr_loop == voice.m_addr_end) + voice.m_enable = false; + } +} + +void gt913_sound_device::data_w(offs_t offset, u16 data) { assert(offset < 3); m_data[offset] = data; } -uint16_t gt913_sound_hle_device::data_r(offs_t offset) +u16 gt913_sound_device::data_r(offs_t offset) { assert(offset < 3); return m_data[offset]; } -void gt913_sound_hle_device::command_w(uint16_t data) +void gt913_sound_device::command_w(u16 data) { - uint8_t voicenum = (data & 0x1f00) >> 8; - uint16_t voicecmd = data & 0x60ff; + m_stream->update(); + + const uint8_t voicenum = (data & 0x1f00) >> 8; + const uint16_t voicecmd = data & 0x60ff; if (data == 0x0012) { - uint8_t gain = m_data[0] & 0x3f; - if (gain != m_gain) - logerror("gain %u\n", gain); - m_gain = gain; + m_gain = m_data[0] & 0x3f; + return; } else if (voicenum >= 24) { return; } - else if (voicecmd == 0x0008) + + auto& voice = m_voices[voicenum]; + if (voicecmd == 0x0008) { /* - Set the voice's sample start point as a ROM address. - This is usually word-aligned, but not always - (e.g. ctk551's lowest piano sample is at address 0x5a801) + sample start addresses seem to need to be word-aligned to decode properly + (see: ctk551 "Trumpet" patch, which will have a bad exponent value otherwise) + this apparently doesn't apply to end/loop addresses, though, or else samples + may loop badly or even become noticeably detuned + TODO: is the LSB of start addresses supposed to indicate something else, then? */ - uint32_t samplestart = (m_data[1] | (m_data[2] << 16)) & 0x1fffff; - logerror("voice %u sample start 0x%06x\n", voicenum, samplestart); + voice.m_addr_start = (m_data[1] | (m_data[2] << 16)) & 0x1ffffe; } else if (voicecmd == 0x0000) { - /* - Set the voice's sample end point as a ROM address. - */ - uint32_t sampleend = (m_data[0] | (m_data[1] << 16)) & 0x1fffff; - logerror("voice %u sample end 0x%06x\n", voicenum, sampleend); + voice.m_addr_end = (m_data[0] | (m_data[1] << 16)) & 0x1fffff; } else if (voicecmd == 0x2000) { - /* - Set the voice's sample loop point as a ROM address. - */ - uint32_t sampleloop = (m_data[0] | (m_data[1] << 16)) & 0x1fffff; - logerror("voice %u sample loop 0x%06x\n", voicenum, sampleloop); + voice.m_addr_loop = (m_data[0] | (m_data[1] << 16)) & 0x1fffff; } else if (voicecmd == 0x200a) { - logerror("voice %u cmd 0x200a (data = %02x)\n", voicenum, m_data[2] & 0xff); + /* TODO: what does bit 4 of data[2] do? ctk551 sets it unconditionally */ + voice.m_exp = m_data[2] & 7; } else if (voicecmd == 0x200b) { - /* - Turn this voice on/off. - ctk551 turns output off before assigning a new note or instrument to this voice, - then turns output back on afterward - */ - logerror("voice %u output %s\n", voicenum, BIT(m_data[2], 7) ? "on" : "off"); + bool enable = BIT(m_data[2], 7); + if (enable && !m_voices[voicenum].m_enable) + { + voice.m_addr_current = voice.m_addr_start; + voice.m_addr_frac = 0; + voice.m_sample = 0; + } + + voice.m_enable = enable; + voice.m_volume_end &= enable; } else if (voicecmd == 0x4004) { - /* - Set this voice's panning, in the form of left and right volume levels (3 bits each) - */ - uint8_t left = (m_data[1] & 0xe0) >> 5; - uint8_t right = (m_data[1] & 0x1c) >> 2; - logerror("voice %u left %u right %u\n", voicenum, left, right); + voice.m_balance[0] = (m_data[1] & 0xe0) >> 5; + voice.m_balance[1] = (m_data[1] & 0x1c) >> 2; } else if (voicecmd == 0x4005) { /* - Set the current pitch of this voice. - The actual format of the value is unknown, but presumably some kind of fixed point + for pitch, data[1] apparently contains both the most and least significant of 4 bytes, + with data0 in the middle. strange, but apparently correct (see higher octaves of ctk551 E.Piano2) */ - uint32_t pitch = (m_data[0] << 8) | (m_data[1] >> 8); - logerror("voice %u pitch 0x%06x\n", voicenum, pitch); + voice.m_pitch = (m_data[1] << 24) | (m_data[0] << 8) | (m_data[1] >> 8); } else if (voicecmd == 0x6006) { - logerror("voice %u cmd 0x6006 (data = %02x)\n", voicenum, m_data[1] & 0xff); + /* + per-voice gain used for normalizing samples + currently treated such that the lower 3 bits are fractional + */ + voice.m_gain = m_data[1] & 0xff; } else if (voicecmd == 0x6007) { + logerror("voice %u volume %u rate %u\n", voicenum, (m_data[0] >> 8), m_data[0] & 0xff); /* - Raise or lower the volume to a specified level at a specified rate. - The actual volume level is probably 7.8 fixed point or something like that, but this command - only sets the most significant bits. + only set a new volume level/rate if we haven't previously indicated the end of an envelope, + unless the new level also has the high bit set. otherwise, a timer irq may try to update the + normal envelope while other code is trying to force a note off */ - logerror("voice %u volume %u rate %u\n", voicenum, (m_data[0] >> 8) & 0x7f, m_data[0] & 0xff); - m_volume_target[voicenum] = m_data[0] & 0x7f00; - m_volume_rate[voicenum] = m_data[0] & 0xff; + const bool end = BIT(m_data[0], 15); + if (!voice.m_volume_end || end) + { + voice.m_volume_end = end; + + voice.m_volume_target = (m_data[0] & 0x7f00) << 16; + /* + In addition to volume levels applying non-linearly, envelope rates + are also non-linear. Unfortunately, with the ctk-551's limited patch set and + lack of editing features, figuring out the correct behavior isn't easy. + This is essentially a rough estimate until a higher-end model (ctk-601 series, etc) + can be dumped and used for more detailed testing. + */ + const u8 x = m_data[0] & 0xff; + if (x >= 127) + voice.m_volume_rate = x << 21; + else if (x >= 63) + voice.m_volume_rate = x << 16; + else if (x >= 47) + voice.m_volume_rate = x << 14; + else if (x >= 31) + voice.m_volume_rate = x << 11; + else if (x >= 23) + voice.m_volume_rate = x << 9; + else if (x >= 15) + voice.m_volume_rate = x << 7; + else + voice.m_volume_rate = x << 5; + } } else if (voicecmd == 0x2028) { /* ctk551 issues this command and then reads the voice's current volume from data0 to determine if it's time to start the next part of the volume envelope or not. - For now, just return the "target" volume immediately - (TODO: also figure out what it expects to be returned in data1) */ - m_data[0] = m_volume_target[voicenum]; - m_data[1] = 0; + m_data[0] = voice.m_enable ? (voice.m_volume_current >> 16) : 0; + /* + data1 is used to read consecutive output sample and detect zero crossings when + applying volume or expression changes to a MIDI channel + */ + m_data[1] = voice.m_sample; } else { @@ -167,9 +346,11 @@ void gt913_sound_hle_device::command_w(uint16_t data) } } -uint16_t gt913_sound_hle_device::status_r() +u16 gt913_sound_device::status_r() { - /* ctk551 reads the current gain level out of the lower 6 bits and ignores the rest - it's unknown what, if anything, the other bits are supposed to contain */ + /* + ctk551 reads the current gain level out of the lower 6 bits and ignores the rest + it's unknown what, if anything, the other bits are supposed to contain + */ return m_gain & 0x3f; } diff --git a/src/devices/machine/gt913_snd.h b/src/devices/machine/gt913_snd.h index b1739fc059d..5e0912ae5d2 100644 --- a/src/devices/machine/gt913_snd.h +++ b/src/devices/machine/gt913_snd.h @@ -9,37 +9,78 @@ #pragma once +#include "dirom.h" + //************************************************************************** // TYPE DEFINITIONS //************************************************************************** -// ======================> gt913_sound_hle_device +// ======================> gt913_sound_device -class gt913_sound_hle_device : public device_t +class gt913_sound_device : public device_t, + public device_sound_interface, + public device_rom_interface<21, 1, 0, ENDIANNESS_BIG> { public: - // construction/destruction - gt913_sound_hle_device(const machine_config &mconfig, const char *tag, device_t *owner, uint32_t clock = 0); + static constexpr feature_type imperfect_features() { return feature::SOUND; } - void data_w(offs_t offset, uint16_t data); - uint16_t data_r(offs_t offset); - void command_w(uint16_t data); - uint16_t status_r(); + // construction/destruction + gt913_sound_device(const machine_config &mconfig, const char *tag, device_t *owner, u32 clock = 0); + + void data_w(offs_t offset, u16 data); + u16 data_r(offs_t offset); + void command_w(u16 data); + u16 status_r(); protected: // device_t overrides virtual void device_start() override; virtual void device_reset() override; -private: - uint8_t m_gain; - uint16_t m_data[3]; + // device_sound_interface overrides + virtual void sound_stream_update(sound_stream &stream, std::vector const &inputs, std::vector &outputs) override; - uint16_t m_volume_target[24]; - uint8_t m_volume_rate[24]; + // device_rom_interface overrides + virtual void rom_bank_updated() override; + +private: + sound_stream *m_stream; + + u8 m_gain; + u16 m_data[3]; + + static const u8 exp_2_to_3[4]; + static const s8 sample_7_to_8[128]; + + struct voice_t + { + bool m_enable; + + u32 m_addr_start; + u32 m_addr_end; + u32 m_addr_loop; + + u32 m_addr_current; + u32 m_addr_frac, m_pitch; + + s16 m_sample, m_sample_next; + u8 m_exp; + + u32 m_volume_current, m_volume_target; + u32 m_volume_rate; + bool m_volume_end; + + u8 m_balance[2]; + u8 m_gain; + }; + + void mix_sample(voice_t& voice, s64& left, s64& right); + void update_sample(voice_t& voice); + + voice_t m_voices[24]; }; // device type definition -DECLARE_DEVICE_TYPE(GT913_SOUND_HLE, gt913_sound_hle_device) +DECLARE_DEVICE_TYPE(GT913_SOUND, gt913_sound_device) #endif // MAME_AUDIO_GT913_H diff --git a/src/mame/drivers/ctk551.cpp b/src/mame/drivers/ctk551.cpp index 77b0859aab8..bb9051be3a9 100644 --- a/src/mame/drivers/ctk551.cpp +++ b/src/mame/drivers/ctk551.cpp @@ -53,6 +53,7 @@ #include "video/hd44780.h" #include "emupal.h" #include "screen.h" +#include "speaker.h" namespace { @@ -64,6 +65,7 @@ public: , m_maincpu(*this, "maincpu") , m_lcdc(*this, "lcdc") , m_led_touch(*this, "led_touch") + , m_led_power(*this, "led_power") { } @@ -90,6 +92,7 @@ private: required_device m_lcdc; output_finder<> m_led_touch; + output_finder<> m_led_power; ioport_value m_switch; }; @@ -97,7 +100,6 @@ private: INPUT_CHANGED_MEMBER(ctk551_state::switch_w) { - logerror("switch_w: %x\n", param); if (!oldval && newval) { if (m_switch == 0x1 && param != m_switch) @@ -112,8 +114,11 @@ INPUT_CHANGED_MEMBER(ctk551_state::switch_w) WRITE_LINE_MEMBER(ctk551_state::apo_w) { logerror("apo_w: %x\n", state); - /* TODO: when 1, this should turn off the LCD, speakers, etc. + /* auto power off - disable the LCD the CPU will go to sleep until the power switch triggers a NMI */ + if (state) + m_lcdc->reset(); + m_led_power = !state; } HD44780_PIXEL_UPDATE(ctk551_state::lcd_update) @@ -140,6 +145,7 @@ void ctk551_state::ctk551_io_map(address_map &map) void ctk551_state::driver_start() { m_led_touch.resolve(); + m_led_power.resolve(); m_switch = 0x2; @@ -149,8 +155,11 @@ void ctk551_state::driver_start() void ctk551_state::ctk551(machine_config &config) { // CPU - GT913(config, m_maincpu, 30'000'000); + // 30MHz oscillator, divided down internally (otherwise the test mode's OK/NG sounds play at double speed) + GT913(config, m_maincpu, 30'000'000 / 2); m_maincpu->set_addrmap(AS_IO, &ctk551_state::ctk551_io_map); + m_maincpu->subdevice("gt_sound")->add_route(0, "lspeaker", 1.0); + m_maincpu->subdevice("gt_sound")->add_route(1, "rspeaker", 1.0); // MIDI auto &mdin(MIDI_PORT(config, "mdin")); @@ -177,6 +186,9 @@ void ctk551_state::ctk551(machine_config &config) screen.set_palette("palette"); PALETTE(config, "palette", FUNC(ctk551_state::palette_init), 2); + + SPEAKER(config, "lspeaker").front_left(); + SPEAKER(config, "rspeaker").front_right(); } INPUT_PORTS_START(ctk551) @@ -356,4 +368,4 @@ ROM_END } // anonymous namespace // YEAR NAME PARENT COMPAT MACHINE INPUT CLASS INIT COMPANY FULLNAME FLAGS -SYST( 1999, ctk551, 0, 0, ctk551, ctk551, ctk551_state, empty_init, "Casio", "CTK-551", MACHINE_NO_SOUND | MACHINE_NOT_WORKING | MACHINE_SUPPORTS_SAVE ) +SYST( 1999, ctk551, 0, 0, ctk551, ctk551, ctk551_state, empty_init, "Casio", "CTK-551", MACHINE_IMPERFECT_GRAPHICS | MACHINE_SUPPORTS_SAVE )