adpcm.c

著名ARC模拟器源码,包括多个平台

/**********************************************************************************************
 *
 *   streaming ADPCM driver
 *   by Aaron Giles
 *
 *   Library to transcode from an ADPCM source to raw PCM.
 *   Written by Buffoni Mirko in 08/06/97
 *   References: various sources and documents.
 *
 *	 HJB 08/31/98
 *	 modified to use an automatically selected oversampling factor
 *	 for the current audio_sample_rate
 *
 *   Mish 21/7/99
 *   Updated to allow multiple OKI chips with different sample rates
 *
 **********************************************************************************************/


#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include "loadroms.h"

#include "driver.h"
#include "adpcm.h"
#include "sasound.h"

#define MAX_SAMPLE_CHUNK	10000

#define FRAC_BITS			14
#define FRAC_ONE			(1 << FRAC_BITS)
#define FRAC_MASK			(FRAC_ONE - 1)


/* struct describing a single playing ADPCM voice */
struct ADPCMVoice
{
	int stream;				/* which stream are we playing on? */
	UINT8 playing;			/* 1 if we are actively playing */

	UINT8 *region_base;		/* pointer to the base of the region */
	UINT8 *base;			/* pointer to the base memory location */
	UINT32 sample;			/* current sample number */
	UINT32 count;			/* total samples to play */

	UINT32 signal;			/* current ADPCM signal */
	UINT32 step;			/* current ADPCM step */
	UINT32 volume;			/* output volume */

	INT16 last_sample;		/* last sample output */
	INT16 curr_sample;		/* current sample target */
	UINT32 source_step;		/* step value for frequency conversion */
	UINT32 source_pos;		/* current fractional position */
};

/* array of ADPCM voices */
static UINT8 num_voices;
static struct ADPCMVoice adpcm[MAX_ADPCM];

/* step size index shift table */
static int index_shift[8] = { -1, -1, -1, -1, 2, 4, 6, 8 };

/* lookup table for the precomputed difference */
static int diff_lookup[49*16];

/* volume lookup table */
static UINT32 volume_table[16];



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

     compute_tables -- compute the difference tables

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

static void compute_tables(void)
{
	/* nibble to bit map */
	static int nbl2bit[16][4] =
	{
		{ 1, 0, 0, 0}, { 1, 0, 0, 1}, { 1, 0, 1, 0}, { 1, 0, 1, 1},
		{ 1, 1, 0, 0}, { 1, 1, 0, 1}, { 1, 1, 1, 0}, { 1, 1, 1, 1},
		{-1, 0, 0, 0}, {-1, 0, 0, 1}, {-1, 0, 1, 0}, {-1, 0, 1, 1},
		{-1, 1, 0, 0}, {-1, 1, 0, 1}, {-1, 1, 1, 0}, {-1, 1, 1, 1}
	};

	int step, nib;

	/* loop over all possible steps */
	for (step = 0; step <= 48; step++)
	{
		/* compute the step value */
		int stepval = floor(16.0 * pow(11.0 / 10.0, (double)step));

		/* loop over all nibbles and compute the difference */
		for (nib = 0; nib < 16; nib++)
		{
			diff_lookup[step*16 + nib] = nbl2bit[nib][0] *
				(stepval   * nbl2bit[nib][1] +
				 stepval/2 * nbl2bit[nib][2] +
				 stepval/4 * nbl2bit[nib][3] +
				 stepval/8);
		}
	}

	/* generate the OKI6295 volume table */
	for (step = 0; step < 16; step++)
	{
		double out = 256.0;
		int vol = step;

		/* 3dB per step */
		while (vol-- > 0)
			out /= 1.412537545;	/* = 10 ^ (3/20) = 3dB */
		volume_table[step] = (UINT32)out;
	}
}



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

     generate_adpcm -- general ADPCM decoding routine

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

static void generate_adpcm(struct ADPCMVoice *voice, INT16 *buffer, int samples)
{
	/* if this voice is active */
	if (voice->playing)
	{
		UINT8 *base = voice->base;
		int sample = voice->sample;
		int signal = voice->signal;
		int count = voice->count;
		int step = voice->step;
		int val;

		/* loop while we still have samples to generate */
		while (samples)
		{
			/* compute the new amplitude and update the current step */
			val = base[sample / 2] >> (((sample & 1) << 2) ^ 4);
			signal += diff_lookup[step * 16 + (val & 15)];

			/* clamp to the maximum */
			if (signal > 2047)
				signal = 2047;
			else if (signal < -2048)
				signal = -2048;

			/* adjust the step size and clamp */
			step += index_shift[val & 7];
			if (step > 48)
				step = 48;
			else if (step < 0)
				step = 0;

			/* output to the buffer, scaling by the volume */
			*buffer++ = signal * voice->volume / 16;
			samples--;

			/* next! */
			if (++sample > count)
			{
				voice->playing = 0;
				break;
			}
		}

		/* update the parameters */
		voice->sample = sample;
		voice->signal = signal;
		voice->step = step;
	}

	/* fill the rest with silence */
	while (samples--)
		*buffer++ = 0;
}



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

     adpcm_update -- update the sound chip so that it is in sync with CPU execution

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

static void adpcm_update(int num, INT16 *buffer, int length)
{
	struct ADPCMVoice *voice = &adpcm[num];
	INT16 sample_data[MAX_SAMPLE_CHUNK], *curr_data = sample_data;
	INT16 prev = voice->last_sample, curr = voice->curr_sample;
	UINT32 final_pos;
	UINT32 new_samples;

	/* finish off the current sample */
	if (voice->source_pos > 0)
	{
		/* interpolate */
		while (length > 0 && voice->source_pos < FRAC_ONE)
		{
			*buffer++ = (((INT32)prev * (FRAC_ONE - voice->source_pos)) + ((INT32)curr * voice->source_pos)) >> FRAC_BITS;
			voice->source_pos += voice->source_step;
			length--;
		}

		/* if we're over, continue; otherwise, we're done */
		if (voice->source_pos >= FRAC_ONE)
			voice->source_pos -= FRAC_ONE;
		else
			return;
	}

	/* compute how many new samples we need */
	final_pos = voice->source_pos + length * voice->source_step;
	new_samples = (final_pos + FRAC_ONE - 1) >> FRAC_BITS;
	if (new_samples > MAX_SAMPLE_CHUNK)
		new_samples = MAX_SAMPLE_CHUNK;

	/* generate them into our buffer */
	generate_adpcm(voice, sample_data, new_samples);
	prev = curr;
	curr = *curr_data++;

	/* then sample-rate convert with linear interpolation */
	while (length > 0)
	{
		/* interpolate */
		while (length > 0 && voice->source_pos < FRAC_ONE)
		{
			*buffer++ = (((INT32)prev * (FRAC_ONE - voice->source_pos)) + ((INT32)curr * voice->source_pos)) >> FRAC_BITS;
			voice->source_pos += voice->source_step;
			length--;
		}

		/* if we're over, grab the next samples */
		if (voice->source_pos >= FRAC_ONE)
		{
			voice->source_pos -= FRAC_ONE;
			prev = curr;
			curr = *curr_data++;
		}
	}

	/* remember the last samples */
	voice->last_sample = prev;
	voice->curr_sample = curr;
}



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

     ADPCM_sh_start -- start emulation of several ADPCM output streams

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

void ADPCMSetBuffers(const struct ADPCMinterface *msound,UINT8 *region,int banksize) {
  // Init 2 chips, or 1 chip with all voices in the same bank.
  int i;
  num_voices = (msound->num * MAX_OKIM6295_VOICES);
  for (i = 0; i < num_voices; i++) {
    adpcm[i].base =
      adpcm[i].region_base = region+(i/MAX_OKIM6295_VOICES)*banksize;
  }
  
}

void ADPCMSetBuffersOne(const struct ADPCMinterface *msound,UINT8 *region,int banksize) {
  // Init 1 chip, 1st bank only (wwfsstars...)
  int i;
  num_voices = (msound->num * MAX_OKIM6295_VOICES);
  for (i = 0; i < num_voices; i++) {
    adpcm[i].base =
      adpcm[i].region_base = region+(i/(MAX_OKIM6295_VOICES/2))*banksize;
  }  
}


static int old_bank;

void OKIM6295_bankswitch(int which, int data) {
  // I know : memcpy is slow... But for now it will do.
  int bank = data & 0xf;
  UINT8 *ADPCM = adpcm[which].region_base;
  if (bank != old_bank) {
    old_bank = bank;
    memcpy(&ADPCM[0x30000], &ADPCM[0x40000 + (bank)*0x10000], 0x10000);
  }
}
  
int ADPCM_sh_start(const struct ADPCMinterface *intf)
{
	char stream_name[40];
	int i;

	/* reset the ADPCM system */
	num_voices = intf->num;
	compute_tables();

	/* initialize the voices */
	memset(adpcm, 0, sizeof(adpcm));
	for (i = 0; i < num_voices; i++)
	{
		/* generate the name and create the stream */
		sprintf(stream_name, "ADPCM #%d", i);
		adpcm[i].stream = stream_init(stream_name, audio_sample_rate, 16,i, adpcm_update);
		/* volume setup */
		stream_set_volume(adpcm[i].stream,intf->mixing_level[0]);		
		if (adpcm[i].stream == -1)
			return 1;

		/* initialize the rest of the structure */
		if (intf->region) // If we have the info...
		  adpcm[i].base = adpcm[i].region_base = load_region[intf->region];
		adpcm[i].volume = 255;
		adpcm[i].signal = -2;
		if (audio_sample_rate)
			adpcm[i].source_step = (UINT32)((double)intf->frequency * (double)FRAC_ONE / (double)audio_sample_rate);
	}

	/* success */
	return 0;
}



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

     ADPCM_sh_stop -- stop emulation of several ADPCM output streams

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

void ADPCM_sh_stop(void)
{
}



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

     ADPCM_sh_update -- update ADPCM streams

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

void ADPCM_sh_update(void)
{
}



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

     ADPCM_play -- play data from a specific offset for a specific length

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

void ADPCM_play(int num, int offset, int length)
{
	struct ADPCMVoice *voice = &adpcm[num];

	/* bail if we're not playing anything */
	if (audio_sample_rate == 0)
		return;

	/* range check the numbers */
	if (num >= num_voices)
	{
#ifdef RAINE_DEBUG
	  fprintf(stderr,"error: ADPCM_trigger() called with channel = %d, but only %d channels allocated\n", num, num_voices);
#endif	  
	  return;
	}

	/* update the ADPCM voice */
	stream_update(voice->stream, 0);
	
	/* set up the voice to play this sample */
	voice->playing = 1;
	voice->base = &voice->region_base[offset];
	voice->sample = 0;
	voice->count = length;

	/* also reset the ADPCM parameters */
	voice->signal = -2;
	voice->step = 0;
}



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

     ADPCM_play -- stop playback on an ADPCM data channel

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

void ADPCM_stop(int num)
{
	struct ADPCMVoice *voice = &adpcm[num];