taito_zm.cpp / zsg2.cpp - more improvements (#3866)

* taito_zm.cpp : Updates Add DSP, Reduce MCFGs, Add device_mixer_interface for sound gain, Add imperfect_features related to DSP, Add notes * taito_zm.cpp : Fix TMS57002 clock * Improve Taito Zoom ZSG-2 sound emulation zsg2.cpp: implement emphasis filter, this is a noise reduction scheme that amplifies higher frequncies to reduce quantization noise. zsg2.cpp: Add sample interpolation and another adjustable lowpass filter. This seems to be roughly what real hardware does... zsg2.cpp: Improve panning registers and identify DSP output gain registers. * zsg2: minor changes [nw] zsg2: Register 0b appears to be status flags [nw] zsg2: Linear ramping probably makes more sense [nw] * zsg2: slight adjustment of emphasis filter [nw] * zsg2: slight adjustment of emphasis filter #2 [nw] * zsg2: more sober ramping algorithm [nw] * tms57002: add instructions 3c/3d, make them behave as NOP as they're undocumented and not understood * tms57002: Add dready callback for superctr (nw) * tms57002: Fixes to make Taito Zoom DSP working tms57002: Add undocumented instruction saom / raom, they set saturation mode for the ALU. tms57002: Implement MACC pipeline. tms57002: Add callbacks for EMPTY and PC0 pins. tms57002: Add a few unimplemented instructions. tms57002: Proper behavior of CMEM UPLOAD mode. tms57002: Fix an issue where program is not properly loaded if PLOAD is set after a program has already been written. * Documentation fix, properly identified registers as ramping control, will implement that soon [nw] * taito_zm: Working DSP emulation Pretty much OST quality now. A pretty decent upgrade from how it was previously, I'd say. * typo [nw] * just adding some quick notes about the WIP [nw] * Fix build [nw] * zsg2: Proper ramping implemenation, add register map, minor cleanups * oops [nw] * taito_zm.cpp / zsg2.cpp - more improvements zsg2.cpp: Attempt to reduce clicks zsg2.cpp: Made the emphasis filter much more simple. I think this matches hardware, as a filter like this could be implemented with very few gates in hardware. Also reset the filter state when the sample position reaches the start address, this fixes raycris song #9 taito_zm.cpp: Adjust volume balance, hopefully fixing psyvarrv. I would need hardware recording for proper verification though tms57002: forgot MACC pipelining for some instructions
2025-04-22 16:31:49 +03:00 · 2018-08-18 22:14:11 +02:00 · 2018-08-18 22:14:11 +02:00 · 3492e6ace4
commit 3492e6ace4
parent c07731b3d4
5 changed files with 114 additions and 68 deletions
--- a/src/devices/cpu/tms57002/tms57002.cpp
+++ b/src/devices/cpu/tms57002/tms57002.cpp
@ -530,7 +530,7 @@ int64_t tms57002_device::macc_to_output_3s(int64_t rounding, uint64_t rmask)

 int64_t tms57002_device::check_macc_overflow_0()
 {
-	int64_t m = macc;
+	int64_t m = macc_read;
 	uint64_t m1;

 	// Overflow detection
@ -543,7 +543,7 @@ int64_t tms57002_device::check_macc_overflow_0()

 int64_t tms57002_device::check_macc_overflow_1()
 {
-	int64_t m = macc;
+	int64_t m = macc_read;
 	uint64_t m1;

 	// Overflow detection
@ -556,7 +556,7 @@ int64_t tms57002_device::check_macc_overflow_1()

 int64_t tms57002_device::check_macc_overflow_2()
 {
-	int64_t m = macc;
+	int64_t m = macc_read;
 	uint64_t m1;

 	// Overflow detection
@ -569,12 +569,12 @@ int64_t tms57002_device::check_macc_overflow_2()

 int64_t tms57002_device::check_macc_overflow_3()
 {
-	return macc;
+	return macc_read;
 }

 int64_t tms57002_device::check_macc_overflow_0s()
 {
-	int64_t m = macc;
+	int64_t m = macc_read;
 	uint64_t m1;

 	// Overflow detection
@ -591,7 +591,7 @@ int64_t tms57002_device::check_macc_overflow_0s()

 int64_t tms57002_device::check_macc_overflow_1s()
 {
-	int64_t m = macc;
+	int64_t m = macc_read;
 	uint64_t m1;

 	// Overflow detection
@ -608,7 +608,7 @@ int64_t tms57002_device::check_macc_overflow_1s()

 int64_t tms57002_device::check_macc_overflow_2s()
 {
-	int64_t m = macc;
+	int64_t m = macc_read;
 	uint64_t m1;

 	// Overflow detection
@ -625,7 +625,35 @@ int64_t tms57002_device::check_macc_overflow_2s()

 int64_t tms57002_device::check_macc_overflow_3s()
 {
-	return macc;
+	return macc_read;
+}
+
+uint32_t tms57002_device::get_cmem(uint8_t addr)
+{
+	if(sa == addr && update_counter_head != update_counter_tail)
+		sti |= S_UPDATE;
+
+	if(sti & S_UPDATE)
+	{
+		cmem[addr] = update[update_counter_tail];
+		update_counter_tail = (update_counter_tail + 1) & 0x0f;
+		update_empty();
+		
+		if(update_counter_head == update_counter_tail)
+			sti &= ~S_UPDATE;
+
+		return cmem[addr]; // The value of crm is ignored during an update.
+	}
+	else
+	{
+		int crm = (st1 & ST1_CRM) >> ST1_CRM_SHIFT;
+		uint32_t cvar = cmem[addr];
+		if(crm == 1)
+			return (cvar & 0xffff0000);
+		else if(crm == 2)
+			return (cvar << 16);
+		return cvar;
+	}
 }

 uint32_t tms57002_device::get_cmem(uint8_t addr)
--- a/src/devices/sound/zsg2.cpp
+++ b/src/devices/sound/zsg2.cpp
@ -24,14 +24,14 @@
 	   +00c : xxxxxxxx xxxxxxxx : End address
 	   +00e : xxxxxxxx -------- : DSP ch 2 (Left) output gain
 	        : -------- xxxxxxxx : Loop address (high)
-	   +010 : xxxxxxxx xxxxxxxx : Filter time constant (latch)
-	   +012 : -------- -------- :   Unknown register (usually cleared)
-	   +014 : xxxxxxxx xxxxxxxx : Volume (latch)
-	   +016 : --x----- -------- : Key on status flag (only read)
-	   +018 : xxxxxxxx xxxxxxxx : Filter time constant (Ramping target)
+	   +010 : xxxxxxxx xxxxxxxx : Initial filter time constant
+	   +012 : xxxxxxxx xxxxxxxx : Current filter time constant
+	   +014 : xxxxxxxx xxxxxxxx : Initial volume
+	   +016 : xxxxxxxx xxxxxxxx : Current volume
+	   +018 : xxxxxxxx xxxxxxxx : Target filter time constant
 	   +01a : xxxxxxxx -------- : DSP ch 1 (chorus) output gain
 	        : -------- xxxxxxxx : Filter ramping speed
-	   +01c : xxxxxxxx xxxxxxxx : Volume (target)
+	   +01c : xxxxxxxx xxxxxxxx : Target volume
 	   +01e : xxxxxxxx -------- : DSP ch 0 (reverb) output gain
 	        : -------- xxxxxxxx : Filter ramping speed
 	600-604 : Key on flags (each bit corresponds to a channel)
@ -97,8 +97,11 @@ TODO:
 #include <fstream>
 #include <cmath>

-#define EMPHASIS_CUTOFF_BASE 0x800
-#define EMPHASIS_OUTPUT_SHIFT 15
+//#define EMPHASIS_CUTOFF_BASE 0x0800
+//#define EMPHASIS_INITIAL_BIAS 0x00400000
+#define EMPHASIS_INITIAL_BIAS 0
+#define EMPHASIS_OUTPUT_SHIFT (16-11)
+#define SAMPLE_OUTPUT_SHIFT 12

 // device type definition
 DEFINE_DEVICE_TYPE(ZSG2, zsg2_device, "zsg2", "ZOOM ZSG-2")
@ -138,13 +141,12 @@ void zsg2_device::device_start()
 	save_item(NAME(m_read_address));

 	// Generate the output gain table. Assuming -1dB per step for now.
-	for (int i = 0, history=0; i < 32; i++)
+	for (int i = 1; i < 32; i++)
 	{
 		double val = pow(10, -(31 - i) / 20.) * 65535.;
-		gain_tab[i] = val;
-		gain_tab_frac[i] = val-history;
-		history = val;
+		m_gain_tab[i] = val;
 	}
+	m_gain_tab[0] = 0;

 	for (int ch = 0; ch < 48; ch++)
 	{
@ -175,7 +177,7 @@ void zsg2_device::device_start()

 		save_item(NAME(m_chan[ch].samples), ch);
 	}
-	
+
 	save_item(NAME(m_sample_count));
 }

@ -204,11 +206,11 @@ void zsg2_device::device_reset()
 	FILE* f;

 	f = fopen("zoom_samples.bin","wb");
-	fwrite(m_mem_copy,1,m_mem_blocks*4,f);
+	fwrite(m_mem_copy.get(),1,m_mem_blocks*4,f);
 	fclose(f);

 	f = fopen("zoom_samples.raw","wb");
-	fwrite(m_full_samples,2,m_mem_blocks*4,f);
+	fwrite(m_full_samples.get(),2,m_mem_blocks*4,f);
 	fclose(f);
 #endif
 }
@ -267,10 +269,8 @@ void zsg2_device::filter_samples(zchan *ch)
 	{
 		ch->samples[i+1] = raw_samples[i];

-		// not sure if the filter works exactly this way, however I am pleased
-		// with the output for now.
-		ch->emphasis_filter_state += (raw_samples[i]-(ch->emphasis_filter_state>>16)) * EMPHASIS_CUTOFF_BASE;
-		ch->samples[i+1] = (ch->emphasis_filter_state) >> EMPHASIS_OUTPUT_SHIFT;
+		ch->emphasis_filter_state += raw_samples[i]-(ch->emphasis_filter_state>>EMPHASIS_OUTPUT_SHIFT);
+		ch->samples[i+1] = ch->emphasis_filter_state >> EMPHASIS_OUTPUT_SHIFT;
 	}
 }

@ -285,20 +285,15 @@ void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **in
 	{
 		int32_t mix[4] = {};

-		int ch = 0;
-
 		// loop over all channels
 		for (auto & elem : m_chan)
-		//auto & elem = m_chan[0];
 		{
-			ch++;
-			if (!(elem.status & STATUS_ACTIVE))
+			if(~elem.status & STATUS_ACTIVE)
 				continue;

 			elem.step_ptr += elem.step;
-			if (elem.step_ptr & 0x10000)
+			if (elem.step_ptr & 0xffff0000)
 			{
-				elem.step_ptr &= 0xffff;
 				if (++elem.cur_pos >= elem.end_pos)
 				{
 					// loop sample
@ -306,26 +301,30 @@ void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **in
 					if ((elem.cur_pos + 1) >= elem.end_pos)
 					{
 						// end of sample
+						elem.vol = 0; //this should help the channel allocation just a bit
 						elem.status &= ~STATUS_ACTIVE;
 						continue;
 					}
+
+					if(elem.cur_pos == elem.start_pos)
+						elem.emphasis_filter_state = EMPHASIS_INITIAL_BIAS;
 				}
+				elem.step_ptr &= 0xffff;
 				filter_samples(&elem);
 			}

 			uint8_t sample_pos = elem.step_ptr >> 14 & 3;
-			int32_t sample; // = elem.samples[sample_pos];
+			int32_t sample = elem.samples[sample_pos];

 			// linear interpolation (hardware certainly does something similar)
-			sample = elem.samples[sample_pos];
 			sample += ((uint16_t)(elem.step_ptr<<2&0xffff) * (int16_t)(elem.samples[sample_pos+1] - sample))>>16;

-			sample = (sample * elem.vol) >> 16;
-
 			// another filter...
 			elem.output_filter_state += (sample - (elem.output_filter_state>>16)) * elem.output_cutoff;
 			sample = elem.output_filter_state >> 16;

+			sample = (sample * elem.vol)>>16;
+
 			for(int output=0; output<4; output++)
 			{
 				int output_gain = elem.output_gain[output] & 0x1f; // left / right
@ -334,7 +333,7 @@ void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **in
 				if (elem.output_gain[output] & 0x80) // perhaps ?
 					output_sample = -output_sample;

-				mix[output] += (output_sample * gain_tab[output_gain&0x1f]) >> 13;
+				mix[output] += (output_sample * m_gain_tab[output_gain&0x1f]) >> SAMPLE_OUTPUT_SHIFT;
 			}

 			// Apply ramping every other update
@ -346,8 +345,6 @@ void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **in
 			}
 		}

-		ch = 0;
-
 		for(int output=0; output<4; output++)
 			outputs[output][i] = mix[output];

@ -379,7 +376,9 @@ void zsg2_device::chan_w(int ch, int reg, uint16_t data)
 			break;

 		case 0x3:
-			// unknown, always 0x0400
+			// unknown, always 0x0400. is this a flag?
+			m_chan[ch].status &= 0x8000;
+			m_chan[ch].status |= data & 0x7fff;
 			break;

 		case 0x4:
@ -412,7 +411,8 @@ void zsg2_device::chan_w(int ch, int reg, uint16_t data)
 			break;

 		case 0x9:
-			// no function? always 0
+			// writes 0 at key on
+			m_chan[ch].output_cutoff = data;
 			break;

 		case 0xa:
@ -421,8 +421,8 @@ void zsg2_device::chan_w(int ch, int reg, uint16_t data)
 			break;

 		case 0xb:
-			// always writes 0
-			// this register is read-only
+			// writes 0 at key on
+			m_chan[ch].vol = data;
 			break;

 		case 0xc:
@ -460,8 +460,16 @@ uint16_t zsg2_device::chan_r(int ch, int reg)
 {
 	switch (reg)
 	{
-		case 0xb: // Only later games (taitogn) read this register...
+		case 0x3:
+			// no games read from this.
 			return m_chan[ch].status;
+		case 0x9:
+			// pretty certain, though no games actually read from this.
+			return m_chan[ch].output_cutoff;
+		case 0xb: // Only later games (taitogn) read this register...
+			// GNet games use some of the flags to decide which channels to kill when
+			// all the channels are busy. (take raycris song #23 as an example)
+			return m_chan[ch].vol;
 		default:
 			break;
 	}
@ -479,14 +487,12 @@ int16_t zsg2_device::get_ramp(uint8_t val)
 {
 	int16_t frac = val<<12; // sign extend
 	frac = ((frac>>12) ^ 8) << (val >> 4);
-	
 	return (frac >> 4);
 }

 inline uint16_t zsg2_device::ramp(uint16_t current, uint16_t target, int16_t delta)
 {
 	int32_t rampval = current + delta;
-	
 	if(delta < 0 && rampval < target)
 		rampval = target;
 	else if(delta >= 0 && rampval > target)
@ -511,12 +517,12 @@ void zsg2_device::control_w(int reg, uint16_t data)
 				{
 					int ch = base | i;
 					m_chan[ch].status |= STATUS_ACTIVE;
-					m_chan[ch].cur_pos = m_chan[ch].start_pos;
-					m_chan[ch].step_ptr = 0;
-					m_chan[ch].emphasis_filter_state = 0;
-					m_chan[ch].vol = m_chan[ch].vol_initial;
+					m_chan[ch].cur_pos = m_chan[ch].start_pos - 1;
+					m_chan[ch].step_ptr = 0x10000;
+					// Ignoring the "initial volume" for now because it causes lots of clicking
+					m_chan[ch].vol = 0; // m_chan[ch].vol_initial;
+					m_chan[ch].vol_delta = 0x0400; // register 06 ?
 					m_chan[ch].output_cutoff = m_chan[ch].output_cutoff_initial;
-					filter_samples(&m_chan[ch]);
 				}
 			}
 			break;
@ -531,6 +537,7 @@ void zsg2_device::control_w(int reg, uint16_t data)
 				if (data & (1 << i))
 				{
 					int ch = base | i;
+					m_chan[ch].vol = 0;
 					m_chan[ch].status &= ~STATUS_ACTIVE;
 				}
 			}
@ -558,6 +565,8 @@ void zsg2_device::control_w(int reg, uint16_t data)
 			break;

 		default:
+			if(reg < 0x20)
+				m_reg[reg] = data;
 			logerror("ZSG2 control   %02X = %04X\n", reg, data & 0xffff);
 			break;
 	}
@ -580,6 +589,8 @@ uint16_t zsg2_device::control_r(int reg)
 			return read_memory(m_read_address) >> 16;

 		default:
+			if(reg < 0x20)
+				return m_reg[reg];
 			break;
 	}

--- a/src/devices/sound/zsg2.h
+++ b/src/devices/sound/zsg2.h
@ -41,13 +41,12 @@ protected:
 	virtual void sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples) override;

 private:
-	const uint16_t STATUS_ACTIVE = 0x2000;
+	const uint16_t STATUS_ACTIVE = 0x8000;

 	// 16 registers per channel, 48 channels
 	struct zchan
 	{
 		uint16_t v[16];
-		
 		uint16_t status;
 		uint32_t cur_pos;
 		uint32_t step_ptr;
@ -68,6 +67,7 @@ private:
 		int16_t output_cutoff_delta;

 		int32_t emphasis_filter_state;
+
 		int32_t output_filter_state;

 		// Attenuation for output channels
@ -76,8 +76,8 @@ private:
 		int16_t samples[5]; // +1 history
 	};

-	uint16_t gain_tab[256];
-	uint16_t gain_tab_frac[256];
+	uint16_t m_gain_tab[256];
+	uint16_t m_reg[32];

 	zchan m_chan[48];
 	uint32_t m_sample_count;
--- a/src/mame/audio/taito_zm.cpp
+++ b/src/mame/audio/taito_zm.cpp
@ -1,5 +1,5 @@
 // license:BSD-3-Clause
-// copyright-holders:Olivier Galibert, hap
+// copyright-holders:Olivier Galibert, hap, superctr, cam900
 /***************************************************************************

    Taito Zoom ZSG-2 sound board
@ -18,16 +18,23 @@ Texas Instruments TMS57002DPHA DSP (QFP80)
 * 1.5625MHz pin 75 and 2 [25/16] (BCKI) (BCKO)

 Newer games have a Panasonic MN1020819DA,
-and a Zoom Corp. ZFX-2 DSP instead of the TMS57002.
-
+and a Zoom Corp. ZFX-2 DSP instead of the TMS57002 (Functionally identical).

 TODO:
+<<<<<<< HEAD
+- ZSG-2 sound chip emulation might not be perfect, see zsg2.cpp
+
+- check DSP behavior. The DSP is programmed to expect 16-bit samples, but
+  the range of the sample values seem really low. Normally the TMS57002 is
+  supposed to left-shift 16-bit samples, but is this the case here?
+=======
 - Raycrisis song 9 gets cut off due to clipping. Possible DSP emulation bug,
  or just have to change the volumes. in the sound test, it will start to cut
  at about 55%. and the timbre doesn't sound right anyway.

 - check DSP behavior
 - Implement the ramping control registers in zsg2.cpp
+>>>>>>> upstream/master

 ***************************************************************************/

@ -209,8 +216,8 @@ MACHINE_CONFIG_START(taito_zoom_device::device_add_mconfig)
 	m_tms57002->empty_callback().set_inputline(m_soundcpu, MN10200_IRQ1).invert();

 	m_tms57002->set_addrmap(AS_DATA, &taito_zoom_device::tms57002_map);
-	m_tms57002->add_route(2, *this, 1.0, AUTO_ALLOC_INPUT, 0);
-	m_tms57002->add_route(3, *this, 1.0, AUTO_ALLOC_INPUT, 1);
+	m_tms57002->add_route(2, *this, 0.5, AUTO_ALLOC_INPUT, 0);
+	m_tms57002->add_route(3, *this, 0.5, AUTO_ALLOC_INPUT, 1);
 #else // Unsupported opcode issue
 	m_tms57002->set_disable();
 #endif
@ -219,11 +226,11 @@ MACHINE_CONFIG_START(taito_zoom_device::device_add_mconfig)
 #ifdef USE_DSP
 	m_zsg2->add_route(0, *m_tms57002, 0.5, 0); // reverb effect
 	m_zsg2->add_route(1, *m_tms57002, 0.5, 1); // chorus effect
-	m_zsg2->add_route(2, *m_tms57002, 0.5, 2); // left direct
-	m_zsg2->add_route(3, *m_tms57002, 0.5, 3); // right direct
+	m_zsg2->add_route(2, *m_tms57002, 1.0, 2); // left direct
+	m_zsg2->add_route(3, *m_tms57002, 1.0, 3); // right direct
 #else
-	m_zsg2->add_route(2, *this, 1.0, AUTO_ALLOC_INPUT, 0);
-	m_zsg2->add_route(3, *this, 1.0, AUTO_ALLOC_INPUT, 1);
+	m_zsg2->add_route(2, *this, 0.5, AUTO_ALLOC_INPUT, 0);
+	m_zsg2->add_route(3, *this, 0.5, AUTO_ALLOC_INPUT, 1);
 #endif

 MACHINE_CONFIG_END
--- a/src/mame/audio/taito_zm.h
+++ b/src/mame/audio/taito_zm.h
@ -1,5 +1,5 @@
 // license:BSD-3-Clause
-// copyright-holders:Olivier Galibert, hap
+// copyright-holders:Olivier Galibert, hap, superctr, cam900
 /***************************************************************************

    Taito Zoom ZSG-2 sound board
@ -14,7 +14,7 @@
 #include "cpu/tms57002/tms57002.h"
 #include "sound/zsg2.h"

-#define USE_DSP               // Uncomment when DSP emulation is working
+#define USE_DSP               // comment out this to disable DSP emulation

 class taito_zoom_device : public device_t, public device_mixer_interface