view src/audio/dc/aica.c @ 942:41a59de7f2ed

Here are patches for SDL12 and SDL_mixer for 4 or 6 channel surround sound on Linux using the Alsa driver. To use them, naturally you need a sound card that will do 4 or 6 channels and probably also a recent version of the Alsa drivers and library. Since the only SDL output driver that knows about surround sound is the Alsa driver, you���ll want to choose it, using: export SDL_AUDIODRIVER=alsa There are no syntactic changes to the programming API. No new library calls, no differences in arguments. There are two semantic changes: (1) For library calls with number of channels as an argument, formerly you could use only 1 or 2 for the number of channels. Now you can also use 4 or 6. (2) The two "left" and "right" arguments to Mix_SetPanning, for the case of 4 or 6 channels, no longer simply control the volumes of the left and right channels. Now the "left" argument is converted to an angle and Mix_SetPosition is called, and the "right" argu- ment is ignored. With two exceptions, so far as I know, the modified SDL12 and SDL_mixer work the same way as the original versions, when opened for 1 or 2 channel output. The two exceptions are bugs which I fixed. Well, the first, anyway, is a bug for sure. When rate conversions up or down by a factor of two are applied (in src/audio/SDL_audiocvt.c), streams with different numbers of channels (that is, mono and stereo) are treated the same way: either each sample is copied or every other sample is omitted. This is ok for mono, but for stereo, it is frames that should be copied or omitted, where by "frame" I mean a portion of the stream containing one sample for each channel. (In the SDL source, confusingly, sometimes frames are called "samples".) So for these rate conversions, stereo streams have to be treated differently, and they are, in my modified version. The other problem that might be characterized as a bug arises when SDL_mixer is passed a multichannel chunk which does not have an integral number of frames. Due to the way the effect_position code loops over frames, when the chunk ends with a partial frame, memory outside the chunk buffer will be accessed. In the case of stereo, it���s possible that because malloc may give more memory than requested, this potential problem never actually causes a segment fault. I don���t know. For 6 channel chunks, I do know, and it does cause segment faults. If SDL_mixer is passed defective chunks and this causes a segment fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not it counts as a bug, it���s easy to protect against, so why not? I added code in mixer.c to discard any partial frame at the end of a chunk. Then what about when SDL or SDL_mixer is opened for 4 or 6 chan- nel output? What happens with the parts of the current library designed for stereo? I don���t know whether I���ve covered all the bases, but I���ve tried: (1) For playing 2 channel waves, or other cases where SDL knows it has to match up a 2 channel source with a 4 or 6 channel output, I���ve added code in SDL_audiocvt.c to make the necessary conversions. (2) For playing midis using timidity, I���ve converted timidity to do 4 or 6 channel output, upon request. (3) For playing mods using mikmod, I put ad hoc code in music.c to convert the stereo output that mikmod produces to 4 or 6 chan- nels. Obviously it would be better to change the mikmod code to mix down into 4 or 6 channels, but I have a hard time following the code in mikmod, so I didn���t do that. (4) For playing mp3s, I put ad hoc code in smpeg to copy channels in the case when 4 or 6 channel output is needed. (5) There seems to be no problem with .ogg files - stereo .oggs can be up converted as .wavs are. (6) The effect_position code in SDL_mixer is now generalized to in- clude the cases of 4 and 6 channel streams. I���ve done a very limited amount of compatibility testing for some of the games using SDL I happen to have. For details, see the file TESTS. I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a couple of 6 channel wave files for testing and a 6 channel ogg file. If you have the right hardware and version of Alsa, you should be able to play the wave files with the Alsa utility aplay (and hear all channels, except maybe lfe, for chan-id.wav, since it���s rather faint). Don���t expect aplay to give good sound, though. There���s something wrong with the current version of aplay. The canyon.ogg file is to test loading of 6 channel oggs. After patching and compiling, you can play it with playmus. (My version of ogg123 will not play it, and I had to patch mplayer to get it to play 6 channel oggs.) Greg Lee <greg@ling.lll.hawaii.edu> Thus, July 1, 2004
author Sam Lantinga <slouken@libsdl.org>
date Sat, 21 Aug 2004 12:27:02 +0000
parents dad72daf44b3
children 11134dc42da8
line wrap: on
line source

/* This file is part of the Dreamcast function library.
 * Please see libdream.c for further details.
 *
 * (c)2000 Dan Potter
 * modify BERO
 */
#include "aica.h"

/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */
#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */

/* Some convienence macros */
#define	SNDREGADDR(x)	(0xa0700000 + (x))
#define	CHNREGADDR(ch,x)	SNDREGADDR(0x80*(ch)+(x))


#define SNDREG32(x)	(*(volatile unsigned long *)SNDREGADDR(x))
#define SNDREG8(x)	(*(volatile unsigned char *)SNDREGADDR(x))
#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x))
#define CHNREG8(ch, x)	(*(volatile unsigned long *)CHNREGADDR(ch,x))

#define G2_LOCK(OLD) \
	do { \
		if (!irq_inside_int()) \
			OLD = irq_disable(); \
		/* suspend any G2 DMA here... */ \
		while((*(volatile unsigned int *)0xa05f688c) & 0x20) \
			; \
	} while(0)

#define G2_UNLOCK(OLD) \
	do { \
		/* resume any G2 DMA here... */ \
		if (!irq_inside_int()) \
			irq_restore(OLD); \
	} while(0)


void aica_init() {
	int i, j, old;
	
	/* Initialize AICA channels */	
	G2_LOCK(old);
	SNDREG32(0x2800) = 0x0000;
	
	for (i=0; i<64; i++) {
		for (j=0; j<0x80; j+=4) {
			if ((j&31)==0) g2_fifo_wait();
			CHNREG32(i, j) = 0;
		}
		g2_fifo_wait();
		CHNREG32(i,0) = 0x8000;
		CHNREG32(i,20) = 0x1f;
	}

	SNDREG32(0x2800) = 0x000f;
	g2_fifo_wait();
	G2_UNLOCK(old);
}

/* Translates a volume from linear form to logarithmic form (required by
   the AICA chip */
/* int logs[] = {

0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103,
105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127,
129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145,
146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159,
160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171,
172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182,
182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191,
191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199,
200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207,
208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215,
215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222,
222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228,
228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234,
234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240,
240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245,
246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251,
251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255

}; */

const static unsigned char logs[] = {
	0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61,
	63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88,
	90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106,
	108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121,
	122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134,
	135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146,
	146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156,
	157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167,
	167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176,
	177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185,
	186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194,
	195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202,
	203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210,
	211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218,
	219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225,
	226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233,
	233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240,
	240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246,
	247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255
};

/* For the moment this is going to have to suffice, until we really
   figure out what these mean. */
#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f)))
#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)])
//#define AICA_VOL(x) (0xff - logs[x&255])

static inline unsigned  AICA_FREQ(unsigned freq)	{
	unsigned long freq_lo, freq_base = 5644800;
	int freq_hi = 7;

	/* Need to convert frequency to floating point format
	   (freq_hi is exponent, freq_lo is mantissa)
	   Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */
	while (freq < freq_base && freq_hi > -8) {
		freq_base >>= 1;
		--freq_hi;
	}
	while (freq < freq_base && freq_hi > -8) {
		freq_base >>= 1;
		freq_hi--;
	}
	freq_lo = (freq<<10) / freq_base;
	return (freq_hi << 11) | (freq_lo & 1023);
}

/* Sets up a sound channel completely. This is generally good if you want
   a quick and dirty way to play notes. If you want a more comprehensive
   set of routines (more like PC wavetable cards) see below.
   
   ch is the channel to play on (0 - 63)
   smpptr is the pointer to the sound data; if you're running off the
     SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just
     ptr. Basically, it's an offset into sound ram.
   mode is one of the mode constants (16 bit, 8 bit, ADPCM)
   nsamp is the number of samples to play (not number of bytes!)
   freq is the sampling rate of the sound
   vol is the volume, 0 to 0xff (0xff is louder)
   pan is a panning constant -- 0 is left, 128 is center, 255 is right.

   This routine (and the similar ones) owe a lot to Marcus' sound example -- 
   I hadn't gotten quite this far into dissecting the individual regs yet. */
void aica_play(int ch,int mode,unsigned long smpptr,int loopst,int loopend,int freq,int vol,int pan,int loopflag) {
	int i;
	int val;
	int old;

	/* Stop the channel (if it's already playing) */
	aica_stop(ch);
	/* doesn't seem to be needed, but it's here just in case */
/*
	for (i=0; i<256; i++) {
		asm("nop");
		asm("nop");
		asm("nop");
		asm("nop");
	}
*/
	G2_LOCK(old);
	/* Envelope setup. The first of these is the loop point,
	   e.g., where the sample starts over when it loops. The second
	   is the loop end. This is the full length of the sample when
	   you are not looping, or the loop end point when you are (though
	   storing more than that is a waste of memory if you're not doing
	   volume enveloping). */
	CHNREG32(ch, 8) = loopst & 0xffff;
	CHNREG32(ch, 12) = loopend & 0xffff;
	
	/* Write resulting values */
	CHNREG32(ch, 24) = AICA_FREQ(freq);
	
	/* Set volume, pan, and some other things that we don't know what
	   they do =) */
	CHNREG32(ch, 36) = AICA_PAN(pan) | (0xf<<8);
	/* Convert the incoming volume and pan into hardware values */
	/* Vol starts at zero so we can ramp */
	vol = AICA_VOL(vol);
	CHNREG32(ch, 40) = 0x24 | (vol<<8);
	/* Convert the incoming volume and pan into hardware values */
	/* Vol starts at zero so we can ramp */

	/* If we supported volume envelopes (which we don't yet) then
	   this value would set that up. The top 4 bits determine the
	   envelope speed. f is the fastest, 1 is the slowest, and 0
	   seems to be an invalid value and does weird things). The
	   default (below) sets it into normal mode (play and terminate/loop).
	CHNREG32(ch, 16) = 0xf010;
	*/
	CHNREG32(ch, 16) = 0x1f;	/* No volume envelope */
	
	
	/* Set sample format, buffer address, and looping control. If
	   0x0200 mask is set on reg 0, the sample loops infinitely. If
	   it's not set, the sample plays once and terminates. We'll
	   also set the bits to start playback here. */
	CHNREG32(ch, 4) = smpptr & 0xffff;
	val = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16);
	if (loopflag) val|=0x200;
	
	CHNREG32(ch, 0) = val;
	
	G2_UNLOCK(old);

	/* Enable playback */
	/* CHNREG32(ch, 0) |= 0xc000; */
	g2_fifo_wait();

#if 0
	for (i=0xff; i>=vol; i--) {
		if ((i&7)==0) g2_fifo_wait();
		CHNREG32(ch, 40) =  0x24 | (i<<8);;
	}

	g2_fifo_wait();
#endif
}

/* Stop the sound on a given channel */
void aica_stop(int ch) {
	g2_write_32(CHNREGADDR(ch, 0),(g2_read_32(CHNREGADDR(ch, 0)) & ~0x4000) | 0x8000);
	g2_fifo_wait();
}


/* The rest of these routines can change the channel in mid-stride so you
   can do things like vibrato and panning effects. */
   
/* Set channel volume */
void aica_vol(int ch,int vol) {
//	g2_write_8(CHNREGADDR(ch, 41),AICA_VOL(vol));
	g2_write_32(CHNREGADDR(ch, 40),(g2_read_32(CHNREGADDR(ch, 40))&0xffff00ff)|(AICA_VOL(vol)<<8) );
	g2_fifo_wait();
}

/* Set channel pan */
void aica_pan(int ch,int pan) {
//	g2_write_8(CHNREGADDR(ch, 36),AICA_PAN(pan));
	g2_write_32(CHNREGADDR(ch, 36),(g2_read_32(CHNREGADDR(ch, 36))&0xffffff00)|(AICA_PAN(pan)) );
	g2_fifo_wait();
}

/* Set channel frequency */
void aica_freq(int ch,int freq) {
	g2_write_32(CHNREGADDR(ch, 24),AICA_FREQ(freq));
	g2_fifo_wait();
}

/* Get channel position */
int aica_get_pos(int ch) {
#if 1
	/* Observe channel ch */
	g2_write_32(SNDREGADDR(0x280c),(g2_read_32(SNDREGADDR(0x280c))&0xffff00ff) | (ch<<8));
	g2_fifo_wait();
	/* Update position counters */
	return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
#else
	/* Observe channel ch */
	g2_write_8(SNDREGADDR(0x280d),ch);
	/* Update position counters */
	return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
#endif
}