Merge of MESS sources (no whatsnew)

src/mess/audio/arcadia.c

@@ -0,0 +1,166 @@
+/***************************************************************************
+
+  PeT mess@utanet.at
+  main part in video/
+
+  Refined with recording/analysis on MPT-03 (PAL UVI chip) by plgDavid
+
+  NTSC UVI sound clock: 15734Hz (arcadia)
+  PAL  UVI sound clock: 15625Hz (Soundic MPT-03 - owned by plgDavid)
+***************************************************************************/
+
+
+#include "includes/arcadia.h"
+
+//known UVI audio clocks
+#define UVI_NTSC 15734
+#define UVI_PAL  15625
+
+/* we need to create pulse transitions that sound 'decent'
+   with the current mess/mame interp scheme
+
+  this is not needed anymore with the new trick in streams.c
+*/
+
+#define OSAMP  1
+
+//lfsr is 9 bits long (and same as Atari TIA pure noise)
+#define LFSR_MASK (1<<8)
+
+//makes alien invaders samples noise sync.
+#define LFSR_INIT 0x00f0
+
+//lfsr states at resynch borders
+//0x01c1
+//0x01e0
+//0x00f0  //good synch
+//0x0178
+//0x01bc
+
+
+// device type definition
+const device_type ARCADIA_SOUND = &device_creator<arcadia_sound_device>;
+
+//-------------------------------------------------
+//  arcadia_sound_device - constructor
+//-------------------------------------------------
+
+arcadia_sound_device::arcadia_sound_device(const machine_config &mconfig, const char *tag, device_t *owner, UINT32 clock)
+	: device_t(mconfig, ARCADIA_SOUND, "Arcadia Custom Sound", tag, owner, clock),
+	  device_sound_interface(mconfig, *this)
+{
+}
+
+//-------------------------------------------------
+//  device_start - device-specific startup
+//-------------------------------------------------
+void arcadia_sound_device::device_start()
+{
+    m_channel = machine().sound().stream_alloc(*this, 0, 1, UVI_PAL*OSAMP, this);
+    m_lfsr    = LFSR_INIT;
+    m_tval    = 1;
+	logerror("arcadia_sound start\n");
+}
+
+//-------------------------------------------------
+//  sound_stream_update - handle update requests for
+//  our sound stream
+//-------------------------------------------------
+
+void arcadia_sound_device::sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples)
+{
+	int i;
+	stream_sample_t *buffer = outputs[0];
+
+	for (i = 0; i < samples; i++, buffer++)
+	{
+		*buffer = 0;
+
+		//if minimal pitch ?
+		if (m_reg[1]){
+			switch (m_mode){
+				//dont play anything
+				case 0:break;
+
+				//tone only
+				case 1:
+					*buffer = m_volume * m_tval;
+				break;
+
+				//noise only
+				case 2:
+					*buffer = m_volume * m_nval;
+				break;
+
+				//tone AND noise (bitwise and)
+				case 3:
+					*buffer = m_volume * (m_tval & m_nval);
+				break;
+			}
+
+			//counter
+			m_pos++;
+
+			if (m_pos >= m_size){
+
+				//calculate new noise bit ( taps: 0000T000T)
+				unsigned char newBit = m_lfsr & 1;         //first tap
+				newBit = (newBit ^ ((m_lfsr & 0x10)?1:0) );//xor with second tap
+
+				m_nval = m_lfsr & 1; //taking new output from LSB
+				m_lfsr = m_lfsr >> 1;//shifting
+
+				//insert new bit at end position (size-1) (only if non null)
+				if (newBit)
+					m_lfsr |= LFSR_MASK;
+
+				//invert tone
+				m_tval = !m_tval;
+
+				m_pos = 0;
+			}
+		}
+	}
+}
+
+
+
+//-------------------------------------------------
+//  soundport_w
+//-------------------------------------------------
+
+WRITE8_MEMBER(arcadia_sound_device::write)
+{
+	m_channel->update();
+	m_reg[offset] = data;
+
+	//logerror("arcadia_sound write:%x=%x\n",offset,data);
+
+	switch (offset)
+	{
+		case 1:
+			//as per Gobbler samples:
+			//the freq counter is only applied on the next change in the flip flop
+			m_size = (data & 0x7f)*OSAMP;
+			//logerror("arcadia_sound write: frq:%d\n",data);
+
+			//reset LFSR
+			if(!m_size)
+				m_lfsr = LFSR_INIT;
+		break;
+
+		case 2:
+			m_volume = (data & 0x07) * 0x800;
+			m_mode   = (data & 0x18) >> 3;
+
+			//logerror("arcadia_sound write: vol:%d mode:%d\n",m_volume,m_mode );
+
+			if (m_mode != m_omode){
+				//not 100% sure about this, maybe we should not reset anything
+				//m_pos  = 0;
+				m_tval = 0;
+			}
+			m_omode = m_mode;
+		break;
+	}
+}

src/mess/audio/arcadia.h

@@ -0,0 +1,41 @@
+#ifndef ARCADIA_SOUND_H_
+#define ARCADIA_SOUND_H_
+
+//**************************************************************************
+//  INTERFACE CONFIGURATION MACROS
+//**************************************************************************
+
+#define MCFG_ARCADIA_SOUND_ADD(_tag) \
+	MCFG_SOUND_ADD(_tag, ARCADIA_SOUND, 0)
+
+//**************************************************************************
+//  TYPE DEFINITIONS
+//**************************************************************************
+
+// ======================> arcadia_sound_device
+
+class arcadia_sound_device : public device_t,
+							 public device_sound_interface
+{
+public:
+	// construction/destruction
+	arcadia_sound_device(const machine_config &mconfig, const char *tag, device_t *owner, UINT32 clock);
+
+	DECLARE_WRITE8_MEMBER(write);
+protected:
+	// device-level overrides
+	virtual void device_start();
+	virtual void sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples);
+
+	sound_stream *m_channel;
+    UINT8 m_reg[3];
+    int m_size, m_pos,m_tval,m_nval;
+	unsigned m_mode, m_omode;
+	unsigned m_volume;
+	unsigned m_lfsr;
+};
+
+// device type definition
+extern const device_type ARCADIA_SOUND;
+
+#endif /* ARCADIA_SOUND_H_ */

src/mess/audio/channelf.c

@@ -0,0 +1,152 @@
+#include "includes/channelf.h"
+
+
+static const int max_amplitude = 0x7fff;
+
+typedef struct _channelf_sound_state channelf_sound_state;
+struct _channelf_sound_state
+{
+	sound_stream *channel;
+	int sound_mode;
+	int incr;
+	float decay_mult;
+	int envelope;
+	UINT32 sample_counter;
+	int forced_ontime;           //  added for improved sound
+	int min_ontime;              //  added for improved sound
+};
+
+INLINE channelf_sound_state *get_safe_token(device_t *device)
+{
+	assert(device != NULL);
+	assert(device->type() == CHANNELF);
+	return (channelf_sound_state *)downcast<legacy_device_base *>(device)->token();
+}
+
+void channelf_sound_w(device_t *device, int mode)
+{
+	channelf_sound_state *state = get_safe_token(device);
+	if (mode == state->sound_mode)
+		return;
+
+	state->channel->update();
+	state->sound_mode = mode;
+
+	switch(mode)
+	{
+		case 0:
+			state->envelope = 0;
+			state->forced_ontime = 0;     //  added for improved sound
+			break;
+		case 1:
+		case 2:
+		case 3:
+			state->envelope = max_amplitude;
+			state->forced_ontime = state->min_ontime;   //  added for improved sound
+			break;
+	}
+}
+
+
+
+static STREAM_UPDATE( channelf_sh_update )
+{
+	channelf_sound_state *state = get_safe_token(device);
+	UINT32 mask = 0, target = 0;
+	stream_sample_t *buffer = outputs[0];
+	stream_sample_t *sample = buffer;
+
+	switch( state->sound_mode )
+	{
+		case 0: /* sound off */
+			memset(buffer,0,sizeof(*buffer)*samples);
+			return;
+
+		case 1: /* high tone (2V) - 1000Hz */
+			mask   = 0x00010000;
+			target = 0x00010000;
+			break;
+		case 2: /* medium tone (4V) - 500Hz */
+			mask   = 0x00020000;
+			target = 0x00020000;
+			break;
+		case 3: /* low (weird) tone (32V & 8V) */
+			mask   = 0x00140000;
+			target = 0x00140000;
+			break;
+	}
+
+	while (samples-- > 0)
+	{
+		if ((state->forced_ontime > 0) || ((state->sample_counter & mask) == target))   //  change made for improved sound
+			*sample++ = state->envelope;
+		else
+			*sample++ = 0;
+		state->sample_counter += state->incr;
+		state->envelope *= state->decay_mult;
+		if (state->forced_ontime > 0)          //  added for improved sound
+			state->forced_ontime -= 1;		//  added for improved sound
+	}
+}
+
+
+
+static DEVICE_START(channelf_sound)
+{
+	channelf_sound_state *state = get_safe_token(device);
+	int rate;
+
+	state->channel = device->machine().sound().stream_alloc(*device, 0, 1, device->machine().sample_rate(), 0, channelf_sh_update);
+	rate = device->machine().sample_rate();
+
+	/*
+     * 2V = 1000Hz ~= 3579535/224/16
+     * Note 2V on the schematic is not the 2V scanline counter -
+     *      it is the 2V vertical pixel counter
+     *      1 pixel = 4 scanlines high
+     *
+     *
+     * This is a convenient way to generate the relevant frequencies,
+     * using a DDS (Direct Digital Synthesizer)
+     *
+     * Essentially, you want a counter to overflow some bit position
+     * at a fixed rate.  So, you figure out a number which you can add
+     * to the counter at every sample, so that you will achieve this
+     *
+     * In this case, we want to overflow bit 16 and the 2V rate, 1000Hz.
+     * This means we also get bit 17 = 4V, bit 18 = 8V, etc.
+     */
+
+	/* This is the proper value to add per sample */
+	state->incr = 65536.0/(rate/1000.0/2.0);
+
+	//  added for improved sound
+	/* This is the minimum forced ontime, in samples */
+	state->min_ontime = rate/1000*2;  /* approx 2ms - estimated, not verified on HW */
+
+	/* This was measured, decay envelope with half life of ~9ms */
+	/* (this is decay multiplier per sample) */
+	state->decay_mult = exp((-0.693/9e-3)/rate);
+
+	/* initial conditions */
+	state->envelope = 0;
+}
+
+
+DEVICE_GET_INFO( channelf_sound )
+{
+	switch (state)
+	{
+		/* --- the following bits of info are returned as 64-bit signed integers --- */
+		case DEVINFO_INT_TOKEN_BYTES:					info->i = sizeof(channelf_sound_state);			break;
+
+		/* --- the following bits of info are returned as pointers to data or functions --- */
+		case DEVINFO_FCT_START:							info->start = DEVICE_START_NAME(channelf_sound);	break;
+
+		/* --- the following bits of info are returned as NULL-terminated strings --- */
+		case DEVINFO_STR_NAME:							strcpy(info->s, "Channel F");				break;
+		case DEVINFO_STR_SOURCE_FILE:					strcpy(info->s, __FILE__);						break;
+	}
+}
+
+DEFINE_LEGACY_SOUND_DEVICE(CHANNELF, channelf_sound);