[Core/Sound] optimized Blip Buffer implementation for stereo streams (thanks to David Knight for original idea)

This commit is contained in:
EkeEke 2016-12-18 23:09:16 +01:00
parent eedbfecac9
commit 41285e1131
12 changed files with 480 additions and 262 deletions

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@ -27,7 +27,7 @@ INCLUDES := core core/m68k core/z80 core/sound core/tremor core/ntsc core/input_
# options for code generation
#---------------------------------------------------------------------------------
CFLAGS = -O3 -fomit-frame-pointer -Wall -Wno-strict-aliasing $(MACHDEP) $(INCLUDE) -DUSE_LIBTREMOR -DDISABLE_MANY_OGG_OPEN_FILES -DUSE_16BPP_RENDERING -DALT_RENDERER
CFLAGS = -O3 -fomit-frame-pointer -Wall -Wno-strict-aliasing $(MACHDEP) $(INCLUDE) -DUSE_LIBTREMOR -DDISABLE_MANY_OGG_OPEN_FILES -DUSE_16BPP_RENDERING -DALT_RENDERER -DBLIP_INVERT
CXXFLAGS = $(CFLAGS)
LDFLAGS = $(MACHDEP) -Wl,-Map,$(notdir $@).map

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@ -27,7 +27,7 @@ INCLUDES := core core/m68k core/z80 core/sound core/tremor core/ntsc core/input_
# options for code generation
#---------------------------------------------------------------------------------
CFLAGS = -O3 -fomit-frame-pointer -Wall -Wno-strict-aliasing $(MACHDEP) $(INCLUDE) -DUSE_LIBTREMOR -DUSE_16BPP_RENDERING -DALT_RENDERER -DHW_RVL
CFLAGS = -O3 -fomit-frame-pointer -Wall -Wno-strict-aliasing $(MACHDEP) $(INCLUDE) -DUSE_LIBTREMOR -DUSE_16BPP_RENDERING -DALT_RENDERER -DBLIP_INVERT -DHW_RVL
CXXFLAGS = $(CFLAGS)
LDFLAGS = $(MACHDEP) -Wl,-Map,$(notdir $@).map,-wrap,wiiuse_set_ir,-wrap,wiiuse_handshake,-wrap,classic_ctrl_handshake,-wrap,classic_ctrl_event

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@ -179,8 +179,7 @@ void cdd_init(int samplerate)
{
/* CD-DA is running by default at 44100 Hz */
/* Audio stream is resampled to desired rate using Blip Buffer */
blip_set_rates(snd.blips[2][0], 44100, samplerate);
blip_set_rates(snd.blips[2][1], 44100, samplerate);
blip_set_rates(snd.blips[2], 44100, samplerate);
}
void cdd_reset(void)
@ -1059,16 +1058,16 @@ void cdd_read_data(uint8 *dst)
void cdd_read_audio(unsigned int samples)
{
/* previous audio outputs */
int16 l = cdd.audio[0];
int16 r = cdd.audio[1];
int prev_l = cdd.audio[0];
int prev_r = cdd.audio[1];
/* get number of internal clocks (samples) needed */
samples = blip_clocks_needed(snd.blips[2][0], samples);
samples = blip_clocks_needed(snd.blips[2], samples);
/* audio track playing ? */
if (!scd.regs[0x36>>1].byte.h && cdd.toc.tracks[cdd.index].fd)
{
int i, mul, delta;
int i, mul, l, r;
/* current CD-DA fader volume */
int curVol = cdd.volume;
@ -1106,17 +1105,13 @@ void cdd_read_audio(unsigned int samples)
/* (MIN) 0,1,2,3,4,8,12,16,20...,1020,1024 (MAX) */
mul = (curVol & 0x7fc) ? (curVol & 0x7fc) : (curVol & 0x03);
/* left channel */
delta = ((ptr[0] * mul) / 1024) - l;
ptr++;
l += delta;
blip_add_delta_fast(snd.blips[2][0], i, delta);
/* right channel */
delta = ((ptr[0] * mul) / 1024) - r;
ptr++;
r += delta;
blip_add_delta_fast(snd.blips[2][1], i, delta);
/* left & right channels */
l = ((ptr[0] * mul) / 1024);
r = ((ptr[1] * mul) / 1024);
blip_add_delta_fast(snd.blips[2], i, l-prev_l, r-prev_r);
prev_l = l;
prev_r = r;
ptr+=2;
/* update CD-DA fader volume (one step/sample) */
if (curVol < endVol)
@ -1153,27 +1148,19 @@ void cdd_read_audio(unsigned int samples)
/* (MIN) 0,1,2,3,4,8,12,16,20...,1020,1024 (MAX) */
mul = (curVol & 0x7fc) ? (curVol & 0x7fc) : (curVol & 0x03);
/* left channel */
/* left & right channels */
#ifdef LSB_FIRST
delta = ((ptr[0] * mul) / 1024) - l;
ptr++;
l = ((ptr[0] * mul) / 1024);
r = ((ptr[1] * mul) / 1024);
ptr+=2;
#else
delta = (((int16)((ptr[0] + ptr[1]*256)) * mul) / 1024) - l;
ptr += 2;
l = (((int16)((ptr[0] + ptr[1]*256)) * mul) / 1024);
r = (((int16)((ptr[2] + ptr[3]*256)) * mul) / 1024);
ptr+=4;
#endif
l += delta;
blip_add_delta_fast(snd.blips[2][0], i, delta);
/* right channel */
#ifdef LSB_FIRST
delta = ((ptr[0] * mul) / 1024) - r;
ptr++;
#else
delta = (((int16)((ptr[0] + ptr[1]*256)) * mul) / 1024) - r;
ptr += 2;
#endif
r += delta;
blip_add_delta_fast(snd.blips[2][1], i, delta);
blip_add_delta_fast(snd.blips[2], i, l-prev_l, r-prev_r);
prev_l = l;
prev_r = r;
/* update CD-DA fader volume (one step/sample) */
if (curVol < endVol)
@ -1198,23 +1185,24 @@ void cdd_read_audio(unsigned int samples)
cdd.volume = curVol;
/* save last audio output for next frame */
cdd.audio[0] = l;
cdd.audio[1] = r;
cdd.audio[0] = prev_l;
cdd.audio[1] = prev_r;
}
else
{
/* no audio output */
if (l) blip_add_delta_fast(snd.blips[2][0], 0, -l);
if (r) blip_add_delta_fast(snd.blips[2][1], 0, -r);
if (prev_l | prev_r)
{
blip_add_delta_fast(snd.blips[2], 0, -prev_l, -prev_r);
/* save audio output for next frame */
cdd.audio[0] = 0;
cdd.audio[1] = 0;
/* save audio output for next frame */
cdd.audio[0] = 0;
cdd.audio[1] = 0;
}
}
/* end of Blip Buffer timeframe */
blip_end_frame(snd.blips[2][0], samples);
blip_end_frame(snd.blips[2][1], samples);
blip_end_frame(snd.blips[2], samples);
}
static void cdd_read_subcode(void)

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@ -2,7 +2,7 @@
* Genesis Plus
* PCM sound chip (315-5476A) (RF5C164 compatible)
*
* Copyright (C) 2012-2014 Eke-Eke (Genesis Plus GX)
* Copyright (C) 2012-2016 Eke-Eke (Genesis Plus GX)
*
* Redistribution and use of this code or any derivative works are permitted
* provided that the following conditions are met:
@ -45,8 +45,7 @@ void pcm_init(double clock, int samplerate)
{
/* PCM chip is running at original rate and is synchronized with SUB-CPU */
/* Chip output is resampled to desired rate using Blip Buffer. */
blip_set_rates(snd.blips[1][0], clock / PCM_SCYCLES_RATIO, samplerate);
blip_set_rates(snd.blips[1][1], clock / PCM_SCYCLES_RATIO, samplerate);
blip_set_rates(snd.blips[1], clock / PCM_SCYCLES_RATIO, samplerate);
}
void pcm_reset(void)
@ -71,8 +70,7 @@ void pcm_reset(void)
pcm.cycles = 0;
/* clear blip buffers */
blip_clear(snd.blips[1][0]);
blip_clear(snd.blips[1][1]);
blip_clear(snd.blips[1]);
}
int pcm_context_save(uint8 *state)
@ -117,6 +115,11 @@ void pcm_run(unsigned int length)
#ifdef LOG_PCM
error("[%d][%d]run %d PCM samples (from %d)\n", v_counter, s68k.cycles, length, pcm.cycles);
#endif
/* previous audio outputs */
int prev_l = pcm.out[0];
int prev_r = pcm.out[1];
/* check if PCM chip is running */
if (pcm.enabled)
{
@ -180,41 +183,29 @@ void pcm_run(unsigned int length)
if (r < -32768) r = -32768;
else if (r > 32767) r = 32767;
/* check if PCM left output changed */
if (pcm.out[0] != l)
{
blip_add_delta_fast(snd.blips[1][0], i, l-pcm.out[0]);
pcm.out[0] = l;
}
/* check if PCM right output changed */
if (pcm.out[1] != r)
{
blip_add_delta_fast(snd.blips[1][1], i, r-pcm.out[1]);
pcm.out[1] = r;
}
/* update Blip Buffer */
blip_add_delta_fast(snd.blips[1], i, l-prev_l, r-prev_r);
prev_l = l;
prev_r = r;
}
/* save last audio outputs */
pcm.out[0] = prev_l;
pcm.out[1] = prev_r;
}
else
{
/* check if PCM left output changed */
if (pcm.out[0])
/* check if PCM output was not muted */
if (prev_l | prev_r)
{
blip_add_delta_fast(snd.blips[1][0], 0, -pcm.out[0]);
blip_add_delta_fast(snd.blips[1], 0, -prev_l, -prev_r);
pcm.out[0] = 0;
}
/* check if PCM right output changed */
if (pcm.out[1])
{
blip_add_delta_fast(snd.blips[1][1], 0, -pcm.out[1]);
pcm.out[1] = 0;
}
}
/* end of blip buffer frame */
blip_end_frame(snd.blips[1][0], length);
blip_end_frame(snd.blips[1][1], length);
blip_end_frame(snd.blips[1], length);
/* update PCM master clock counter */
pcm.cycles += length * PCM_SCYCLES_RATIO;
@ -223,7 +214,7 @@ void pcm_run(unsigned int length)
void pcm_update(unsigned int samples)
{
/* get number of internal clocks (samples) needed */
unsigned int clocks = blip_clocks_needed(snd.blips[1][0], samples);
unsigned int clocks = blip_clocks_needed(snd.blips[1], samples);
/* run PCM chip */
if (clocks > 0)

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@ -2,7 +2,7 @@
* Genesis Plus
* PCM sound chip (315-5476A) (RF5C164 compatible)
*
* Copyright (C) 2012-2014 Eke-Eke (Genesis Plus GX)
* Copyright (C) 2012-2016 Eke-Eke (Genesis Plus GX)
*
* Redistribution and use of this code or any derivative works are permitted
* provided that the following conditions are met:

View File

@ -1,10 +1,11 @@
/* blip_buf $vers. http://www.slack.net/~ant/ */
/* blip_buf $vers. http://www.slack.net/~ant/ */
/* Modified for Genesis Plus GX by EkeEke (01/09/12) */
/* - disabled assertions checks (define #BLIP_ASSERT to re-enable) */
/* - fixed multiple time-frames support & removed m->avail */
/* - modified blip_read_samples to always output to stereo streams */
/* - added blip_mix_samples function (see blip_buf.h) */
/* Modified for Genesis Plus GX by EkeEke */
/* - disabled assertions checks (define #BLIP_ASSERT to re-enable) */
/* - fixed multiple time-frames support & removed m->avail */
/* - added blip_mix_samples function (see blip_buf.h) */
/* - added stereo buffer support (define #BLIP_MONO to disable) */
/* - added inverted stereo output (define #BLIP_INVERT to enable)*/
#include "blip_buf.h"
@ -61,24 +62,32 @@ enum { phase_count = 1 << phase_bits };
enum { delta_bits = 15 };
enum { delta_unit = 1 << delta_bits };
enum { frac_bits = time_bits - pre_shift };
enum { phase_shift = frac_bits - phase_bits };
/* We could eliminate avail and encode whole samples in offset, but that would
limit the total buffered samples to blip_max_frame. That could only be
increased by decreasing time_bits, which would reduce resample ratio accuracy.
*/
typedef int buf_t;
struct blip_t
{
fixed_t factor;
fixed_t offset;
int size;
#ifdef BLIP_MONO
int integrator;
#else
int integrator[2];
buf_t* buffer[2];
#endif
};
typedef int buf_t;
#ifdef BLIP_MONO
/* probably not totally portable */
#define SAMPLES( buf ) ((buf_t*) ((buf) + 1))
#define SAMPLES( blip ) ((buf_t*) ((blip) + 1))
#endif
/* Arithmetic (sign-preserving) right shift */
#define ARITH_SHIFT( n, shift ) \
@ -124,9 +133,23 @@ blip_t* blip_new( int size )
assert( size >= 0 );
#endif
#ifdef BLIP_MONO
m = (blip_t*) malloc( sizeof *m + (size + buf_extra) * sizeof (buf_t) );
#else
m = (blip_t*) malloc( sizeof *m );
#endif
if ( m )
{
#ifndef BLIP_MONO
m->buffer[0] = (buf_t*) malloc( (size + buf_extra) * sizeof (buf_t));
m->buffer[1] = (buf_t*) malloc( (size + buf_extra) * sizeof (buf_t));
if ((m->buffer[0] == NULL) || (m->buffer[1] == NULL))
{
blip_delete(m);
return 0;
}
#endif
m->factor = time_unit / blip_max_ratio;
m->size = size;
blip_clear( m );
@ -141,7 +164,13 @@ void blip_delete( blip_t* m )
{
if ( m != NULL )
{
/* Clear fields in case user tries to use after freeing */
#ifndef BLIP_MONO
if (m->buffer[0] != NULL)
free(m->buffer[0]);
if (m->buffer[1] != NULL)
free(m->buffer[1]);
#endif
/* Clear fields in case user tries to use after freeing */
memset( m, 0, sizeof *m );
free( m );
}
@ -174,9 +203,16 @@ void blip_clear( blip_t* m )
with the slight loss of showing an error in half the time. Since for
a 64-bit factor this is years, the halving isn't a problem. */
m->offset = m->factor / 2;
m->offset = m->factor / 2;
#ifdef BLIP_MONO
m->integrator = 0;
memset( SAMPLES( m ), 0, (m->size + buf_extra) * sizeof (buf_t) );
#else
m->integrator[0] = 0;
m->integrator[1] = 0;
memset( m->buffer[0], 0, (m->size + buf_extra) * sizeof (buf_t) );
memset( m->buffer[1], 0, (m->size + buf_extra) * sizeof (buf_t) );
#endif
}
int blip_clocks_needed( const blip_t* m, int samples )
@ -212,12 +248,21 @@ int blip_samples_avail( const blip_t* m )
static void remove_samples( blip_t* m, int count )
{
#ifdef BLIP_MONO
buf_t* buf = SAMPLES( m );
int remain = (m->offset >> time_bits) + buf_extra - count;
#else
buf_t* buf = m->buffer[0];
#endif
int remain = (m->offset >> time_bits) + buf_extra - count;
m->offset -= count * time_unit;
memmove( &buf [0], &buf [count], remain * sizeof buf [0] );
memset( &buf [remain], 0, count * sizeof buf [0] );
memmove( &buf [0], &buf [count], remain * sizeof (buf_t) );
memset( &buf [remain], 0, count * sizeof (buf_t) );
#ifndef BLIP_MONO
buf = m->buffer[1];
memmove( &buf [0], &buf [count], remain * sizeof (buf_t) );
memset( &buf [remain], 0, count * sizeof (buf_t) );
#endif
}
int blip_read_samples( blip_t* m, short out [], int count)
@ -231,9 +276,16 @@ int blip_read_samples( blip_t* m, short out [], int count)
if ( count )
#endif
{
buf_t const* in = SAMPLES( m );
buf_t const* end = in + count;
#ifdef BLIP_MONO
buf_t const* in = SAMPLES( m );
int sum = m->integrator;
#else
buf_t const* in = m->buffer[0];
buf_t const* in2 = m->buffer[1];
int sum = m->integrator[0];
int sum2 = m->integrator[1];
#endif
buf_t const* end = in + count;
do
{
/* Eliminate fraction */
@ -243,60 +295,120 @@ int blip_read_samples( blip_t* m, short out [], int count)
CLAMP( s );
*out = s;
out += 2;
*out++ = s;
/* High-pass filter */
sum -= s << (delta_bits - bass_shift);
#ifndef BLIP_MONO
/* Eliminate fraction */
s = ARITH_SHIFT( sum2, delta_bits );
sum2 += *in2++;
CLAMP( s );
*out++ = s;
/* High-pass filter */
sum2 -= s << (delta_bits - bass_shift);
#endif
}
while ( in != end );
m->integrator = sum;
#ifdef BLIP_MONO
m->integrator = sum;
#else
m->integrator[0] = sum;
m->integrator[1] = sum2;
#endif
remove_samples( m, count );
}
return count;
}
int blip_mix_samples( blip_t* m, short out [], int count)
int blip_mix_samples( blip_t* m1, blip_t* m2, blip_t* m3, short out [], int count)
{
#ifdef BLIP_ASSERT
assert( count >= 0 );
assert( count >= 0 );
if ( count > (m->offset >> time_bits) )
count = m->offset >> time_bits;
if ( count > (m1->offset >> time_bits) )
count = m1->offset >> time_bits;
if ( count > (m2->offset >> time_bits) )
count = m2->offset >> time_bits;
if ( count > (m3->offset >> time_bits) )
count = m3->offset >> time_bits;
if ( count )
if ( count )
#endif
{
buf_t const* in = SAMPLES( m );
buf_t const* end = in + count;
int sum = m->integrator;
do
{
/* Eliminate fraction */
int s = ARITH_SHIFT( sum, delta_bits );
buf_t const* end;
buf_t const* in[3];
#ifdef BLIP_MONO
int sum = m1->integrator;
in[0] = SAMPLES( m1 );
in[1] = SAMPLES( m2 );
in[2] = SAMPLES( m3 );
#else
int sum = m1->integrator[0];
int sum2 = m1->integrator[1];
buf_t const* in2[3];
in[0] = m1->buffer[0];
in[1] = m2->buffer[0];
in[2] = m3->buffer[0];
in2[0] = m1->buffer[1];
in2[1] = m2->buffer[1];
in2[2] = m3->buffer[1];
#endif
sum += *in++;
end = in[0] + count;
do
{
/* Eliminate fraction */
int s = ARITH_SHIFT( sum, delta_bits );
/* High-pass filter */
sum -= s << (delta_bits - bass_shift);
sum += *in[0]++;
sum += *in[1]++;
sum += *in[2]++;
/* Add current buffer value */
s += *out;
CLAMP( s );
CLAMP( s );
*out++ = s;
*out = s;
out += 2;
}
while ( in != end );
m->integrator = sum;
/* High-pass filter */
sum -= s << (delta_bits - bass_shift);
remove_samples( m, count );
}
#ifndef BLIP_MONO
/* Eliminate fraction */
s = ARITH_SHIFT( sum2, delta_bits );
return count;
sum2 += *in2[0]++;
sum2 += *in2[1]++;
sum2 += *in2[2]++;
CLAMP( s );
*out++ = s;
/* High-pass filter */
sum2 -= s << (delta_bits - bass_shift);
#endif
}
while ( in[0] != end );
#ifdef BLIP_MONO
m1->integrator = sum;
#else
m1->integrator[0] = sum;
m1->integrator[1] = sum2;
#endif
remove_samples( m1, count );
remove_samples( m2, count );
remove_samples( m3, count );
}
return count;
}
/* Things that didn't help performance on x86:
@ -348,12 +460,180 @@ possibly-wider fixed_t. On 32-bit platforms, this is likely more efficient.
And by having pre_shift 32, a 32-bit platform can easily do the shift by
simply ignoring the low half. */
#ifndef BLIP_MONO
void blip_add_delta( blip_t* m, unsigned time, int delta_l, int delta_r )
{
if (delta_l | delta_r)
{
unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
int phase = fixed >> phase_shift & (phase_count - 1);
short const* in = bl_step [phase];
short const* rev = bl_step [phase_count - phase];
int interp = fixed >> (phase_shift - delta_bits) & (delta_unit - 1);
int pos = fixed >> frac_bits;
#ifdef BLIP_INVERT
buf_t* out_l = m->buffer[1] + pos;
buf_t* out_r = m->buffer[0] + pos;
#else
buf_t* out_l = m->buffer[0] + pos;
buf_t* out_r = m->buffer[1] + pos;
#endif
int delta;
#ifdef BLIP_ASSERT
/* Fails if buffer size was exceeded */
assert( pos <= m->size + end_frame_extra );
#endif
if (delta_l == delta_r)
{
buf_t out;
delta = (delta_l * interp) >> delta_bits;
delta_l -= delta;
out = in[0]*delta_l + in[half_width+0]*delta;
out_l[0] += out;
out_r[0] += out;
out = in[1]*delta_l + in[half_width+1]*delta;
out_l[1] += out;
out_r[1] += out;
out = in[2]*delta_l + in[half_width+2]*delta;
out_l[2] += out;
out_r[2] += out;
out = in[3]*delta_l + in[half_width+3]*delta;
out_l[3] += out;
out_r[3] += out;
out = in[4]*delta_l + in[half_width+4]*delta;
out_l[4] += out;
out_r[4] += out;
out = in[5]*delta_l + in[half_width+5]*delta;
out_l[5] += out;
out_r[5] += out;
out = in[6]*delta_l + in[half_width+6]*delta;
out_l[6] += out;
out_r[6] += out;
out = in[7]*delta_l + in[half_width+7]*delta;
out_l[7] += out;
out_r[7] += out;
out = rev[7]*delta_l + rev[7-half_width]*delta;
out_l[8] += out;
out_r[8] += out;
out = rev[6]*delta_l + rev[6-half_width]*delta;
out_l[9] += out;
out_r[9] += out;
out = rev[5]*delta_l + rev[5-half_width]*delta;
out_l[10] += out;
out_r[10] += out;
out = rev[4]*delta_l + rev[4-half_width]*delta;
out_l[11] += out;
out_r[11] += out;
out = rev[3]*delta_l + rev[3-half_width]*delta;
out_l[12] += out;
out_r[12] += out;
out = rev[2]*delta_l + rev[2-half_width]*delta;
out_l[13] += out;
out_r[13] += out;
out = rev[1]*delta_l + rev[1-half_width]*delta;
out_l[14] += out;
out_r[14] += out;
out = rev[0]*delta_l + rev[0-half_width]*delta;
out_l[15] += out;
out_r[15] += out;
}
else
{
delta = (delta_l * interp) >> delta_bits;
delta_l -= delta;
out_l [0] += in[0]*delta_l + in[half_width+0]*delta;
out_l [1] += in[1]*delta_l + in[half_width+1]*delta;
out_l [2] += in[2]*delta_l + in[half_width+2]*delta;
out_l [3] += in[3]*delta_l + in[half_width+3]*delta;
out_l [4] += in[4]*delta_l + in[half_width+4]*delta;
out_l [5] += in[5]*delta_l + in[half_width+5]*delta;
out_l [6] += in[6]*delta_l + in[half_width+6]*delta;
out_l [7] += in[7]*delta_l + in[half_width+7]*delta;
out_l [8] += rev[7]*delta_l + rev[7-half_width]*delta;
out_l [9] += rev[6]*delta_l + rev[6-half_width]*delta;
out_l [10] += rev[5]*delta_l + rev[5-half_width]*delta;
out_l [11] += rev[4]*delta_l + rev[4-half_width]*delta;
out_l [12] += rev[3]*delta_l + rev[3-half_width]*delta;
out_l [13] += rev[2]*delta_l + rev[2-half_width]*delta;
out_l [14] += rev[1]*delta_l + rev[1-half_width]*delta;
out_l [15] += rev[0]*delta_l + rev[0-half_width]*delta;
delta = (delta_r * interp) >> delta_bits;
delta_r -= delta;
out_r [0] += in[0]*delta_r + in[half_width+0]*delta;
out_r [1] += in[1]*delta_r + in[half_width+1]*delta;
out_r [2] += in[2]*delta_r + in[half_width+2]*delta;
out_r [3] += in[3]*delta_r + in[half_width+3]*delta;
out_r [4] += in[4]*delta_r + in[half_width+4]*delta;
out_r [5] += in[5]*delta_r + in[half_width+5]*delta;
out_r [6] += in[6]*delta_r + in[half_width+6]*delta;
out_r [7] += in[7]*delta_r + in[half_width+7]*delta;
out_r [8] += rev[7]*delta_r + rev[7-half_width]*delta;
out_r [9] += rev[6]*delta_r + rev[6-half_width]*delta;
out_r [10] += rev[5]*delta_r + rev[5-half_width]*delta;
out_r [11] += rev[4]*delta_r + rev[4-half_width]*delta;
out_r [12] += rev[3]*delta_r + rev[3-half_width]*delta;
out_r [13] += rev[2]*delta_r + rev[2-half_width]*delta;
out_r [14] += rev[1]*delta_r + rev[1-half_width]*delta;
out_r [15] += rev[0]*delta_r + rev[0-half_width]*delta;
}
}
}
void blip_add_delta_fast( blip_t* m, unsigned time, int delta_l, int delta_r )
{
if (delta_l | delta_r)
{
unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
int interp = fixed >> (frac_bits - delta_bits) & (delta_unit - 1);
int pos = fixed >> frac_bits;
#ifdef STEREO_INVERT
buf_t* out_l = m->buffer[1] + pos;
buf_t* out_r = m->buffer[0] + pos;
#else
buf_t* out_l = m->buffer[0] + pos;
buf_t* out_r = m->buffer[1] + pos;
#endif
int delta = delta_l * interp;
#ifdef BLIP_ASSERT
/* Fails if buffer size was exceeded */
assert( pos <= m->size + end_frame_extra );
#endif
if (delta_l == delta_r)
{
delta_l = delta_l * delta_unit - delta;
out_l[7] += delta_l;
out_l[8] += delta;
out_r[7] += delta_l;
out_r[8] += delta;
}
else
{
out_l[7] += delta_l * delta_unit - delta;
out_l[8] += delta;
delta = delta_r * interp;
out_r[7] += delta_r * delta_unit - delta;
out_r[8] += delta;
}
}
}
#else
void blip_add_delta( blip_t* m, unsigned time, int delta )
{
unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
buf_t* out = SAMPLES( m ) + (fixed >> frac_bits);
int const phase_shift = frac_bits - phase_bits;
int phase = fixed >> phase_shift & (phase_count - 1);
short const* in = bl_step [phase];
short const* rev = bl_step [phase_count - phase];
@ -403,3 +683,4 @@ void blip_add_delta_fast( blip_t* m, unsigned time, int delta )
out [7] += delta * delta_unit - delta2;
out [8] += delta2;
}
#endif

View File

@ -28,12 +28,24 @@ blip_max_ratio = 1 << 20 };
/** Clears entire buffer. Afterwards, blip_samples_avail() == 0. */
void blip_clear( blip_t* );
#ifndef BLIP_MONO
/** Adds positive/negative deltas into stereo buffers at specified clock time. */
void blip_add_delta( blip_t*, unsigned time, int delta_l, int delta_r );
/** Same as blip_add_delta(), but uses faster, lower-quality synthesis. */
void blip_add_delta_fast( blip_t*, unsigned int clock_time, int delta_l, int delta_r );
#else
/** Adds positive/negative delta into buffer at specified clock time. */
void blip_add_delta( blip_t*, unsigned int clock_time, int delta );
/** Same as blip_add_delta(), but uses faster, lower-quality synthesis. */
void blip_add_delta_fast( blip_t*, unsigned int clock_time, int delta );
#endif
/** Length of time frame, in clocks, needed to make sample_count additional
samples available. */
int blip_clocks_needed( const blip_t*, int sample_count );
@ -56,9 +68,8 @@ element of 'out', allowing easy interleaving of two buffers into a stereo sample
stream. Outputs 16-bit signed samples. Returns number of samples actually read. */
int blip_read_samples( blip_t*, short out [], int count);
/* Same as above function except sample is added to output buffer previous value */
/* This allows easy mixing of different blip buffers into a single output stream */
int blip_mix_samples( blip_t* m, short out [], int count);
/* Same as above function except sample is mixed from three blip buffers source */
int blip_mix_samples( blip_t* m1, blip_t* m2, blip_t* m3, short out [], int count);
/** Frees buffer. No effect if NULL is passed. */
void blip_delete( blip_t* );

View File

@ -167,45 +167,23 @@ int SN76489_GetContextSize(void)
/* Updates tone amplitude in delta buffer. Call whenever amplitude might have changed. */
INLINE void UpdateToneAmplitude(int i, int time)
{
int delta;
/* left output */
delta = (SN76489.Channel[i][0] * SN76489.ToneFreqPos[i]) - SN76489.ChanOut[i][0];
if (delta != 0)
{
SN76489.ChanOut[i][0] += delta;
blip_add_delta(snd.blips[0][0], time, delta);
}
/* right output */
delta = (SN76489.Channel[i][1] * SN76489.ToneFreqPos[i]) - SN76489.ChanOut[i][1];
if (delta != 0)
{
SN76489.ChanOut[i][1] += delta;
blip_add_delta(snd.blips[0][1], time, delta);
}
/* left & right output */
int delta_l = (SN76489.Channel[i][0] * SN76489.ToneFreqPos[i]) - SN76489.ChanOut[i][0];
int delta_r = (SN76489.Channel[i][1] * SN76489.ToneFreqPos[i]) - SN76489.ChanOut[i][1];
blip_add_delta(snd.blips[0], time, delta_l, delta_r);
SN76489.ChanOut[i][0] += delta_l;
SN76489.ChanOut[i][1] += delta_r;
}
/* Updates noise amplitude in delta buffer. Call whenever amplitude might have changed. */
INLINE void UpdateNoiseAmplitude(int time)
{
int delta;
/* left output */
delta = (SN76489.Channel[3][0] * ( SN76489.NoiseShiftRegister & 0x1 )) - SN76489.ChanOut[3][0];
if (delta != 0)
{
SN76489.ChanOut[3][0] += delta;
blip_add_delta(snd.blips[0][0], time, delta);
}
/* right output */
delta = (SN76489.Channel[3][1] * ( SN76489.NoiseShiftRegister & 0x1 )) - SN76489.ChanOut[3][1];
if (delta != 0)
{
SN76489.ChanOut[3][1] += delta;
blip_add_delta(snd.blips[0][1], time, delta);
}
/* left & right output */
int delta_l = (SN76489.Channel[3][0] * ( SN76489.NoiseShiftRegister & 0x1 )) - SN76489.ChanOut[3][0];
int delta_r = (SN76489.Channel[3][1] * ( SN76489.NoiseShiftRegister & 0x1 )) - SN76489.ChanOut[3][1];
blip_add_delta(snd.blips[0], time, delta_l, delta_r);
SN76489.ChanOut[3][0] += delta_l;
SN76489.ChanOut[3][1] += delta_r;
}
/* Runs tone channel for clock_length clocks */

View File

@ -2,8 +2,8 @@
* Genesis Plus
* Sound Hardware
*
* Copyright (C) 1998, 1999, 2000, 2001, 2002, 2003 Charles Mac Donald (original code)
* Copyright (C) 2007-2013 Eke-Eke (Genesis Plus GX)
* Copyright (C) 1998-2003 Charles Mac Donald (original code)
* Copyright (C) 2007-2016 Eke-Eke (Genesis Plus GX)
*
* Redistribution and use of this code or any derivative works are permitted
* provided that the following conditions are met:
@ -125,21 +125,21 @@ void sound_reset(void)
int sound_update(unsigned int cycles)
{
int delta, preamp, time, l, r, *ptr;
int prev_l, prev_r, preamp, time, l, r, *ptr;
/* Run PSG & FM chips until end of frame */
SN76489_Update(cycles);
fm_update(cycles);
/* FM output pre-amplification */
/* FM output pre-amplification */
preamp = config.fm_preamp;
/* FM frame initial timestamp */
time = fm_cycles_start;
/* Restore last FM outputs from previous frame */
l = fm_last[0];
r = fm_last[1];
prev_l = fm_last[0];
prev_r = fm_last[1];
/* FM buffer start pointer */
ptr = fm_buffer;
@ -150,15 +150,12 @@ int sound_update(unsigned int cycles)
/* high-quality Band-Limited synthesis */
do
{
/* left channel */
delta = ((*ptr++ * preamp) / 100) - l;
l += delta;
blip_add_delta(snd.blips[0][0], time, delta);
/* right channel */
delta = ((*ptr++ * preamp) / 100) - r;
r += delta;
blip_add_delta(snd.blips[0][1], time, delta);
/* left & right channels */
l = ((*ptr++ * preamp) / 100);
r = ((*ptr++ * preamp) / 100);
blip_add_delta(snd.blips[0], time, l-prev_l, r-prev_r);
prev_l = l;
prev_r = r;
/* increment time counter */
time += fm_cycles_ratio;
@ -170,15 +167,12 @@ int sound_update(unsigned int cycles)
/* faster Linear Interpolation */
do
{
/* left channel */
delta = ((*ptr++ * preamp) / 100) - l;
l += delta;
blip_add_delta_fast(snd.blips[0][0], time, delta);
/* right channel */
delta = ((*ptr++ * preamp) / 100) - r;
r += delta;
blip_add_delta_fast(snd.blips[0][1], time, delta);
/* left & right channels */
l = ((*ptr++ * preamp) / 100);
r = ((*ptr++ * preamp) / 100);
blip_add_delta_fast(snd.blips[0], time, l-prev_l, r-prev_r);
prev_l = l;
prev_r = r;
/* increment time counter */
time += fm_cycles_ratio;
@ -190,18 +184,17 @@ int sound_update(unsigned int cycles)
fm_ptr = fm_buffer;
/* save last FM output for next frame */
fm_last[0] = l;
fm_last[1] = r;
fm_last[0] = prev_l;
fm_last[1] = prev_r;
/* adjust FM cycle counters for next frame */
fm_cycles_count = fm_cycles_start = time - cycles;
/* end of blip buffers time frame */
blip_end_frame(snd.blips[0][0], cycles);
blip_end_frame(snd.blips[0][1], cycles);
/* end of blip buffer time frame */
blip_end_frame(snd.blips[0], cycles);
/* return number of available samples */
return blip_samples_avail(snd.blips[0][0]);
return blip_samples_avail(snd.blips[0]);
}
int sound_context_save(uint8 *state)

View File

@ -2,8 +2,8 @@
* Genesis Plus
* Sound Hardware
*
* Copyright (C) 1998, 1999, 2000, 2001, 2002, 2003 Charles Mac Donald (original code)
* Copyright (C) 2007-2013 Eke-Eke (Genesis Plus GX)
* Copyright (C) 1998-2003 Charles Mac Donald (original code)
* Copyright (C) 2007-2016 Eke-Eke (Genesis Plus GX)
*
* Redistribution and use of this code or any derivative works are permitted
* provided that the following conditions are met:

View File

@ -68,11 +68,9 @@ int audio_init(int samplerate, double framerate)
memset(&snd, 0, sizeof (snd));
/* Initialize Blip Buffers */
snd.blips[0][0] = blip_new(samplerate / 10);
snd.blips[0][1] = blip_new(samplerate / 10);
if (!snd.blips[0][0] || !snd.blips[0][1])
snd.blips[0] = blip_new(samplerate / 10);
if (!snd.blips[0])
{
audio_shutdown();
return -1;
}
@ -80,11 +78,9 @@ int audio_init(int samplerate, double framerate)
if (system_hw == SYSTEM_MCD)
{
/* allocate blip buffers */
snd.blips[1][0] = blip_new(samplerate / 10);
snd.blips[1][1] = blip_new(samplerate / 10);
snd.blips[2][0] = blip_new(samplerate / 10);
snd.blips[2][1] = blip_new(samplerate / 10);
if (!snd.blips[1][0] || !snd.blips[1][1] || !snd.blips[2][0] || !snd.blips[2][1])
snd.blips[1] = blip_new(samplerate / 10);
snd.blips[2] = blip_new(samplerate / 10);
if (!snd.blips[1] || !snd.blips[2])
{
audio_shutdown();
return -1;
@ -132,8 +128,7 @@ void audio_set_rate(int samplerate, double framerate)
/* master clock timebase so they remain perfectly synchronized together, while still */
/* being synchronized with 68K and Z80 CPUs as well. Mixed sound chip output is then */
/* resampled to desired rate at the end of each frame, using Blip Buffer. */
blip_set_rates(snd.blips[0][0], mclk, samplerate);
blip_set_rates(snd.blips[0][1], mclk, samplerate);
blip_set_rates(snd.blips[0], mclk, samplerate);
/* Mega CD sound hardware */
if (system_hw == SYSTEM_MCD)
@ -155,17 +150,14 @@ void audio_set_rate(int samplerate, double framerate)
void audio_reset(void)
{
int i,j;
int i;
/* Clear blip buffers */
for (i=0; i<3; i++)
{
for (j=0; j<2; j++)
if (snd.blips[i])
{
if (snd.blips[i][j])
{
blip_clear(snd.blips[i][j]);
}
blip_clear(snd.blips[i]);
}
}
@ -187,16 +179,13 @@ void audio_set_equalizer(void)
void audio_shutdown(void)
{
int i,j;
int i;
/* Delete blip buffers */
for (i=0; i<3; i++)
{
for (j=0; j<2; j++)
{
blip_delete(snd.blips[i][j]);
snd.blips[i][j] = 0;
}
blip_delete(snd.blips[i]);
snd.blips[i] = 0;
}
}
@ -213,37 +202,24 @@ int audio_update(int16 *buffer)
/* read CDDA samples */
cdd_read_audio(size);
}
#ifdef ALIGN_SND
/* return an aligned number of samples if required */
size &= ALIGN_SND;
/* return an aligned number of samples if required */
size &= ALIGN_SND;
#endif
/* resample FM & PSG mixed stream to output buffer */
#ifdef LSB_FIRST
blip_read_samples(snd.blips[0][0], buffer, size);
blip_read_samples(snd.blips[0][1], buffer + 1, size);
#else
blip_read_samples(snd.blips[0][0], buffer + 1, size);
blip_read_samples(snd.blips[0][1], buffer, size);
#endif
/* Mega CD specific */
if (system_hw == SYSTEM_MCD)
/* resample & mix FM/PSG, PCM & CD-DA streams to output buffer */
blip_mix_samples(snd.blips[0], snd.blips[1], snd.blips[2], buffer, size);
}
else
{
/* resample PCM & CD-DA streams to output buffer */
#ifdef LSB_FIRST
blip_mix_samples(snd.blips[1][0], buffer, size);
blip_mix_samples(snd.blips[1][1], buffer + 1, size);
blip_mix_samples(snd.blips[2][0], buffer, size);
blip_mix_samples(snd.blips[2][1], buffer + 1, size);
#else
blip_mix_samples(snd.blips[1][0], buffer + 1, size);
blip_mix_samples(snd.blips[1][1], buffer, size);
blip_mix_samples(snd.blips[2][0], buffer + 1, size);
blip_mix_samples(snd.blips[2][1], buffer, size);
#ifdef ALIGN_SND
/* return an aligned number of samples if required */
size &= ALIGN_SND;
#endif
/* resample FM/PSG mixed stream to output buffer */
blip_read_samples(snd.blips[0], buffer, size);
}
/* Audio filtering */

View File

@ -91,7 +91,7 @@ typedef struct
int sample_rate; /* Output Sample rate (8000-48000) */
double frame_rate; /* Output Frame rate (usually 50 or 60 frames per second) */
int enabled; /* 1= sound emulation is enabled */
blip_t* blips[3][2]; /* Blip Buffer resampling */
blip_t* blips[3]; /* Blip Buffer resampling (stereo) */
} t_snd;