diff options
| author | PCSX* teams | 2010-11-16 14:15:22 +0200 |
|---|---|---|
| committer | Grazvydas Ignotas | 2010-11-16 14:15:22 +0200 |
| commit | ef79bbde537d6b9c745a7d86cb9df1d04c35590d (patch) | |
| tree | ef8d2520dbb9e1e345b41b12c9959f300ca8fd10 /plugins/dfsound | |
| download | pcsx_rearmed-ef79bbde537d6b9c745a7d86cb9df1d04c35590d.tar.gz pcsx_rearmed-ef79bbde537d6b9c745a7d86cb9df1d04c35590d.tar.bz2 pcsx_rearmed-ef79bbde537d6b9c745a7d86cb9df1d04c35590d.zip | |
pcsxr-1.9.92
Diffstat (limited to 'plugins/dfsound')
30 files changed, 6601 insertions, 0 deletions
diff --git a/plugins/dfsound/Makefile.am b/plugins/dfsound/Makefile.am new file mode 100644 index 0000000..88a7dc8 --- /dev/null +++ b/plugins/dfsound/Makefile.am @@ -0,0 +1,51 @@ +INCLUDES = -DPIXMAPDIR=\"${datadir}/pixmaps/\" \ + -DLOCALE_DIR=\"${datadir}/locale/\" \ + -DDATADIR=\"${datadir}/psemu/\" \ + $(GTK2_CFLAGS) $(GLADE2_CFLAGS) \ + -I../../include + +bindir = @libdir@/games/psemu/ +libdir = @libdir@/games/psemu/ + +lib_LTLIBRARIES = libDFSound.la + +libDFSound_la_SOURCES = spu.c cfg.c dma.c freeze.c registers.c + +libDFSound_la_CFLAGS = +libDFSound_la_LDFLAGS = -module -avoid-version -lpthread -lm + +if SOUND_ALSA +libDFSound_la_SOURCES += alsa.c +libDFSound_la_CFLAGS += -DUSEALSA=1 +libDFSound_la_LDFLAGS += $(ALSA_LIBS) +endif + +if SOUND_OSS +libDFSound_la_SOURCES += oss.c +libDFSound_la_CFLAGS += -DUSEOSS=1 +endif + +if SOUND_PULSEAUDIO +libDFSound_la_SOURCES += pulseaudio.c +libDFSound_la_CFLAGS += -DUSEPULSEAUDIO=1 $(PULSEAUDIO_CFLAGS) +libDFSound_la_LDFLAGS += $(PULSEAUDIO_LIBS) +endif + +if SOUND_SDL +libDFSound_la_SOURCES += sdl.c +libDFSound_la_CFLAGS += -DUSESDL=1 $(SDL_CFLAGS) +libDFSound_la_LDFLAGS += $(SDL_LIBS) +endif + +if SOUND_NULL +libDFSound_la_SOURCES += nullsnd.c +libDFSound_la_CFLAGS += -DUSENULL=1 +endif + +bin_PROGRAMS = cfgDFSound +cfgDFSound_SOURCES = spucfg-0.1df/main.c +cfgDFSound_LDADD = $(GTK2_LIBS) $(GLADE2_LIBS) + +glade_DATA = spucfg-0.1df/dfsound.glade2 +gladedir = $(datadir)/psemu/ +EXTRA_DIST = $(glade_DATA) diff --git a/plugins/dfsound/Makefile.in b/plugins/dfsound/Makefile.in new file mode 100644 index 0000000..70edf5b --- /dev/null +++ b/plugins/dfsound/Makefile.in @@ -0,0 +1,711 @@ +# Makefile.in generated by automake 1.10.2 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002, +# 2003, 2004, 2005, 2006, 2007, 2008 Free Software Foundation, Inc. +# This Makefile.in is free software; the Free Software Foundation +# gives unlimited permission to copy and/or distribute it, +# with or without modifications, as long as this notice is preserved. + +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY, to the extent permitted by law; without +# even the implied warranty of MERCHANTABILITY or FITNESS FOR A +# PARTICULAR PURPOSE. + +@SET_MAKE@ + + + +VPATH = @srcdir@ +pkgdatadir = $(datadir)/@PACKAGE@ +pkglibdir = $(libdir)/@PACKAGE@ +pkgincludedir = $(includedir)/@PACKAGE@ +am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd +install_sh_DATA = $(install_sh) -c -m 644 +install_sh_PROGRAM = $(install_sh) -c +install_sh_SCRIPT = $(install_sh) -c +INSTALL_HEADER = $(INSTALL_DATA) +transform = $(program_transform_name) +NORMAL_INSTALL = : +PRE_INSTALL = : +POST_INSTALL = : +NORMAL_UNINSTALL = : +PRE_UNINSTALL = : +POST_UNINSTALL = : +build_triplet = @build@ +host_triplet = @host@ +target_triplet = @target@ +@SOUND_ALSA_TRUE@am__append_1 = alsa.c +@SOUND_ALSA_TRUE@am__append_2 = -DUSEALSA=1 +@SOUND_ALSA_TRUE@am__append_3 = $(ALSA_LIBS) +@SOUND_OSS_TRUE@am__append_4 = oss.c +@SOUND_OSS_TRUE@am__append_5 = -DUSEOSS=1 +@SOUND_PULSEAUDIO_TRUE@am__append_6 = pulseaudio.c +@SOUND_PULSEAUDIO_TRUE@am__append_7 = -DUSEPULSEAUDIO=1 $(PULSEAUDIO_CFLAGS) +@SOUND_PULSEAUDIO_TRUE@am__append_8 = $(PULSEAUDIO_LIBS) +@SOUND_SDL_TRUE@am__append_9 = sdl.c +@SOUND_SDL_TRUE@am__append_10 = -DUSESDL=1 $(SDL_CFLAGS) +@SOUND_SDL_TRUE@am__append_11 = $(SDL_LIBS) +@SOUND_NULL_TRUE@am__append_12 = nullsnd.c +@SOUND_NULL_TRUE@am__append_13 = -DUSENULL=1 +bin_PROGRAMS = cfgDFSound$(EXEEXT) +subdir = plugins/dfsound +DIST_COMMON = $(srcdir)/Makefile.am $(srcdir)/Makefile.in +ACLOCAL_M4 = $(top_srcdir)/aclocal.m4 +am__aclocal_m4_deps = $(top_srcdir)/configure.ac +am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \ + $(ACLOCAL_M4) +mkinstalldirs = $(SHELL) $(top_srcdir)/mkinstalldirs +CONFIG_HEADER = $(top_builddir)/include/config.h +CONFIG_CLEAN_FILES = +am__vpath_adj_setup = srcdirstrip=`echo "$(srcdir)" | sed 's|.|.|g'`; +am__vpath_adj = case $$p in \ + $(srcdir)/*) f=`echo "$$p" | sed "s|^$$srcdirstrip/||"`;; \ + *) f=$$p;; \ + esac; +am__strip_dir = `echo $$p | sed -e 's|^.*/||'`; +am__installdirs = "$(DESTDIR)$(libdir)" "$(DESTDIR)$(bindir)" \ + "$(DESTDIR)$(gladedir)" +libLTLIBRARIES_INSTALL = $(INSTALL) +LTLIBRARIES = $(lib_LTLIBRARIES) +libDFSound_la_LIBADD = +am__libDFSound_la_SOURCES_DIST = spu.c cfg.c dma.c freeze.c \ + registers.c alsa.c oss.c pulseaudio.c sdl.c nullsnd.c +@SOUND_ALSA_TRUE@am__objects_1 = libDFSound_la-alsa.lo +@SOUND_OSS_TRUE@am__objects_2 = libDFSound_la-oss.lo +@SOUND_PULSEAUDIO_TRUE@am__objects_3 = libDFSound_la-pulseaudio.lo +@SOUND_SDL_TRUE@am__objects_4 = libDFSound_la-sdl.lo +@SOUND_NULL_TRUE@am__objects_5 = libDFSound_la-nullsnd.lo +am_libDFSound_la_OBJECTS = libDFSound_la-spu.lo libDFSound_la-cfg.lo \ + libDFSound_la-dma.lo libDFSound_la-freeze.lo \ + libDFSound_la-registers.lo $(am__objects_1) $(am__objects_2) \ + $(am__objects_3) $(am__objects_4) $(am__objects_5) +libDFSound_la_OBJECTS = $(am_libDFSound_la_OBJECTS) +libDFSound_la_LINK = $(LIBTOOL) --tag=CC $(AM_LIBTOOLFLAGS) \ + $(LIBTOOLFLAGS) --mode=link $(CCLD) $(libDFSound_la_CFLAGS) \ + $(CFLAGS) $(libDFSound_la_LDFLAGS) $(LDFLAGS) -o $@ +binPROGRAMS_INSTALL = $(INSTALL_PROGRAM) +PROGRAMS = $(bin_PROGRAMS) +am_cfgDFSound_OBJECTS = main.$(OBJEXT) +cfgDFSound_OBJECTS = $(am_cfgDFSound_OBJECTS) +am__DEPENDENCIES_1 = +cfgDFSound_DEPENDENCIES = $(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) +DEFAULT_INCLUDES = -I.@am__isrc@ -I$(top_builddir)/include +depcomp = $(SHELL) $(top_srcdir)/depcomp +am__depfiles_maybe = depfiles +COMPILE = $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) \ + $(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS) +LTCOMPILE = $(LIBTOOL) --tag=CC $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) \ + --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) \ + $(AM_CPPFLAGS) $(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS) +CCLD = $(CC) +LINK = $(LIBTOOL) --tag=CC $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) \ + --mode=link $(CCLD) $(AM_CFLAGS) $(CFLAGS) $(AM_LDFLAGS) \ + $(LDFLAGS) -o $@ +SOURCES = $(libDFSound_la_SOURCES) $(cfgDFSound_SOURCES) +DIST_SOURCES = $(am__libDFSound_la_SOURCES_DIST) $(cfgDFSound_SOURCES) +gladeDATA_INSTALL = $(INSTALL_DATA) +DATA = $(glade_DATA) +ETAGS = etags +CTAGS = ctags +DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST) +ACLOCAL = @ACLOCAL@ +ALSA_CFLAGS = @ALSA_CFLAGS@ +ALSA_LIBS = @ALSA_LIBS@ +AMTAR = @AMTAR@ +AR = @AR@ +AUTOCONF = @AUTOCONF@ +AUTOHEADER = @AUTOHEADER@ +AUTOMAKE = @AUTOMAKE@ +AWK = @AWK@ +CC = @CC@ +CCAS = @CCAS@ +CCASDEPMODE = @CCASDEPMODE@ +CCASFLAGS = @CCASFLAGS@ +CCDEPMODE = @CCDEPMODE@ +CFLAGS = @CFLAGS@ +CPP = @CPP@ +CPPFLAGS = @CPPFLAGS@ +CYGPATH_W = @CYGPATH_W@ +DEFS = @DEFS@ +DEPDIR = @DEPDIR@ +DSYMUTIL = @DSYMUTIL@ +DUMPBIN = @DUMPBIN@ +ECHO_C = @ECHO_C@ +ECHO_N = @ECHO_N@ +ECHO_T = @ECHO_T@ +EGREP = @EGREP@ +EXEEXT = @EXEEXT@ +FGREP = @FGREP@ +GETTEXT_MACRO_VERSION 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/dev/null +++ b/plugins/dfsound/adsr.c @@ -0,0 +1,641 @@ +/***************************************************************************
+ adsr.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_ADSR
+
+// will be included from spu.c
+#ifdef _IN_SPU
+
+////////////////////////////////////////////////////////////////////////
+// ADSR func
+////////////////////////////////////////////////////////////////////////
+
+unsigned long RateTable[160];
+
+void InitADSR(void) // INIT ADSR
+{
+ unsigned long r,rs,rd;int i;
+
+ memset(RateTable,0,sizeof(unsigned long)*160); // build the rate table according to Neill's rules (see at bottom of file)
+
+ r=3;rs=1;rd=0;
+
+ for(i=32;i<160;i++) // we start at pos 32 with the real values... everything before is 0
+ {
+ if(r<0x3FFFFFFF)
+ {
+ r+=rs;
+ rd++;if(rd==5) {rd=1;rs*=2;}
+ }
+ if(r>0x3FFFFFFF) r=0x3FFFFFFF;
+
+ RateTable[i]=r;
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE void StartADSR(int ch) // MIX ADSR
+{
+ s_chan[ch].ADSRX.lVolume=1; // and init some adsr vars
+ s_chan[ch].ADSRX.State=0;
+ s_chan[ch].ADSRX.EnvelopeVol=0;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int MixADSR(int ch) // MIX ADSR
+{
+ if(s_chan[ch].bStop) // should be stopped:
+ { // do release
+ if(s_chan[ch].ADSRX.ReleaseModeExp)
+ {
+ switch((s_chan[ch].ADSRX.EnvelopeVol>>28)&0x7)
+ {
+ case 0: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +0 + 32]; break;
+ case 1: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +4 + 32]; break;
+ case 2: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +6 + 32]; break;
+ case 3: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +8 + 32]; break;
+ case 4: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +9 + 32]; break;
+ case 5: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +10+ 32]; break;
+ case 6: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +11+ 32]; break;
+ case 7: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x18 +12+ 32]; break;
+ }
+ }
+ else
+ {
+ s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.ReleaseRate^0x1F))-0x0C + 32];
+ }
+
+ if(s_chan[ch].ADSRX.EnvelopeVol<0)
+ {
+ s_chan[ch].ADSRX.EnvelopeVol=0;
+ s_chan[ch].bOn=0;
+ //s_chan[ch].bReverb=0;
+ //s_chan[ch].bNoise=0;
+ }
+
+ s_chan[ch].ADSRX.lVolume=s_chan[ch].ADSRX.EnvelopeVol>>21;
+ return s_chan[ch].ADSRX.lVolume;
+ }
+ else // not stopped yet?
+ {
+ if(s_chan[ch].ADSRX.State==0) // -> attack
+ {
+ if(s_chan[ch].ADSRX.AttackModeExp)
+ {
+ if(s_chan[ch].ADSRX.EnvelopeVol<0x60000000)
+ s_chan[ch].ADSRX.EnvelopeVol+=RateTable[(s_chan[ch].ADSRX.AttackRate^0x7F)-0x10 + 32];
+ else
+ s_chan[ch].ADSRX.EnvelopeVol+=RateTable[(s_chan[ch].ADSRX.AttackRate^0x7F)-0x18 + 32];
+ }
+ else
+ {
+ s_chan[ch].ADSRX.EnvelopeVol+=RateTable[(s_chan[ch].ADSRX.AttackRate^0x7F)-0x10 + 32];
+ }
+
+ if(s_chan[ch].ADSRX.EnvelopeVol<0)
+ {
+ s_chan[ch].ADSRX.EnvelopeVol=0x7FFFFFFF;
+ s_chan[ch].ADSRX.State=1;
+ }
+
+ s_chan[ch].ADSRX.lVolume=s_chan[ch].ADSRX.EnvelopeVol>>21;
+ return s_chan[ch].ADSRX.lVolume;
+ }
+ //--------------------------------------------------//
+ if(s_chan[ch].ADSRX.State==1) // -> decay
+ {
+ switch((s_chan[ch].ADSRX.EnvelopeVol>>28)&0x7)
+ {
+ case 0: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+0 + 32]; break;
+ case 1: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+4 + 32]; break;
+ case 2: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+6 + 32]; break;
+ case 3: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+8 + 32]; break;
+ case 4: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+9 + 32]; break;
+ case 5: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+10+ 32]; break;
+ case 6: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+11+ 32]; break;
+ case 7: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[(4*(s_chan[ch].ADSRX.DecayRate^0x1F))-0x18+12+ 32]; break;
+ }
+
+ if(s_chan[ch].ADSRX.EnvelopeVol<0) s_chan[ch].ADSRX.EnvelopeVol=0;
+ if(((s_chan[ch].ADSRX.EnvelopeVol>>27)&0xF) <= s_chan[ch].ADSRX.SustainLevel)
+ {
+ s_chan[ch].ADSRX.State=2;
+ }
+
+ s_chan[ch].ADSRX.lVolume=s_chan[ch].ADSRX.EnvelopeVol>>21;
+ return s_chan[ch].ADSRX.lVolume;
+ }
+ //--------------------------------------------------//
+ if(s_chan[ch].ADSRX.State==2) // -> sustain
+ {
+ if(s_chan[ch].ADSRX.SustainIncrease)
+ {
+ if(s_chan[ch].ADSRX.SustainModeExp)
+ {
+ if(s_chan[ch].ADSRX.EnvelopeVol<0x60000000)
+ s_chan[ch].ADSRX.EnvelopeVol+=RateTable[(s_chan[ch].ADSRX.SustainRate^0x7F)-0x10 + 32];
+ else
+ s_chan[ch].ADSRX.EnvelopeVol+=RateTable[(s_chan[ch].ADSRX.SustainRate^0x7F)-0x18 + 32];
+ }
+ else
+ {
+ s_chan[ch].ADSRX.EnvelopeVol+=RateTable[(s_chan[ch].ADSRX.SustainRate^0x7F)-0x10 + 32];
+ }
+
+ if(s_chan[ch].ADSRX.EnvelopeVol<0)
+ {
+ s_chan[ch].ADSRX.EnvelopeVol=0x7FFFFFFF;
+ }
+ }
+ else
+ {
+ if(s_chan[ch].ADSRX.SustainModeExp)
+ {
+ switch((s_chan[ch].ADSRX.EnvelopeVol>>28)&0x7)
+ {
+ case 0: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +0 + 32];break;
+ case 1: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +4 + 32];break;
+ case 2: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +6 + 32];break;
+ case 3: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +8 + 32];break;
+ case 4: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +9 + 32];break;
+ case 5: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +10+ 32];break;
+ case 6: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +11+ 32];break;
+ case 7: s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x1B +12+ 32];break;
+ }
+ }
+ else
+ {
+ s_chan[ch].ADSRX.EnvelopeVol-=RateTable[((s_chan[ch].ADSRX.SustainRate^0x7F))-0x0F + 32];
+ }
+
+ if(s_chan[ch].ADSRX.EnvelopeVol<0)
+ {
+ s_chan[ch].ADSRX.EnvelopeVol=0;
+ }
+ }
+ s_chan[ch].ADSRX.lVolume=s_chan[ch].ADSRX.EnvelopeVol>>21;
+ return s_chan[ch].ADSRX.lVolume;
+ }
+ }
+ return 0;
+}
+
+#endif
+
+/*
+James Higgs ADSR investigations:
+
+PSX SPU Envelope Timings
+~~~~~~~~~~~~~~~~~~~~~~~~
+
+First, here is an extract from doomed's SPU doc, which explains the basics
+of the SPU "volume envelope":
+
+*** doomed doc extract start ***
+
+--------------------------------------------------------------------------
+Voices.
+--------------------------------------------------------------------------
+The SPU has 24 hardware voices. These voices can be used to reproduce sample
+data, noise or can be used as frequency modulator on the next voice.
+Each voice has it's own programmable ADSR envelope filter. The main volume
+can be programmed independently for left and right output.
+
+The ADSR envelope filter works as follows:
+Ar = Attack rate, which specifies the speed at which the volume increases
+ from zero to it's maximum value, as soon as the note on is given. The
+ slope can be set to lineair or exponential.
+Dr = Decay rate specifies the speed at which the volume decreases to the
+ sustain level. Decay is always decreasing exponentially.
+Sl = Sustain level, base level from which sustain starts.
+Sr = Sustain rate is the rate at which the volume of the sustained note
+ increases or decreases. This can be either lineair or exponential.
+Rr = Release rate is the rate at which the volume of the note decreases
+ as soon as the note off is given.
+
+ lvl |
+ ^ | /\Dr __
+ Sl _| _ / _ \__--- \
+ | / ---__ \ Rr
+ | /Ar Sr \ \
+ | / \\
+ |/___________________\________
+ ->time
+
+The overal volume can also be set to sweep up or down lineairly or
+exponentially from it's current value. This can be done seperately
+for left and right.
+
+Relevant SPU registers:
+-------------------------------------------------------------
+$1f801xx8 Attack/Decay/Sustain level
+bit |0f|0e 0d 0c 0b 0a 09 08|07 06 05 04|03 02 01 00|
+desc.|Am| Ar |Dr |Sl |
+
+Am 0 Attack mode Linear
+ 1 Exponential
+
+Ar 0-7f attack rate
+Dr 0-f decay rate
+Sl 0-f sustain level
+-------------------------------------------------------------
+$1f801xxa Sustain rate, Release Rate.
+bit |0f|0e|0d|0c 0b 0a 09 08 07 06|05|04 03 02 01 00|
+desc.|Sm|Sd| 0| Sr |Rm|Rr |
+
+Sm 0 sustain rate mode linear
+ 1 exponential
+Sd 0 sustain rate mode increase
+ 1 decrease
+Sr 0-7f Sustain Rate
+Rm 0 Linear decrease
+ 1 Exponential decrease
+Rr 0-1f Release Rate
+
+Note: decay mode is always Expontial decrease, and thus cannot
+be set.
+-------------------------------------------------------------
+$1f801xxc Current ADSR volume
+bit |0f 0e 0d 0c 0b 0a 09 08 07 06 05 04 03 02 01 00|
+desc.|ADSRvol |
+
+ADSRvol Returns the current envelope volume when
+ read.
+-- James' Note: return range: 0 -> 32767
+
+*** doomed doc extract end ***
+
+By using a small PSX proggie to visualise the envelope as it was played,
+the following results for envelope timing were obtained:
+
+1. Attack rate value (linear mode)
+
+ Attack value range: 0 -> 127
+
+ Value | 48 | 52 | 56 | 60 | 64 | 68 | 72 | | 80 |
+ -----------------------------------------------------------------
+ Frames | 11 | 21 | 42 | 84 | 169| 338| 676| |2890|
+
+ Note: frames is no. of PAL frames to reach full volume (100%
+ amplitude)
+
+ Hmm, noticing that the time taken to reach full volume doubles
+ every time we add 4 to our attack value, we know the equation is
+ of form:
+ frames = k * 2 ^ (value / 4)
+
+ (You may ponder about envelope generator hardware at this point,
+ or maybe not... :)
+
+ By substituting some stuff and running some checks, we get:
+
+ k = 0.00257 (close enuf)
+
+ therefore,
+ frames = 0.00257 * 2 ^ (value / 4)
+ If you just happen to be writing an emulator, then you can probably
+ use an equation like:
+
+ %volume_increase_per_tick = 1 / frames
+
+
+ ------------------------------------
+ Pete:
+ ms=((1<<(value>>2))*514)/10000
+ ------------------------------------
+
+2. Decay rate value (only has log mode)
+
+ Decay value range: 0 -> 15
+
+ Value | 8 | 9 | 10 | 11 | 12 | 13 | 14 | 15 |
+ ------------------------------------------------
+ frames | | | | | 6 | 12 | 24 | 47 |
+
+ Note: frames here is no. of PAL frames to decay to 50% volume.
+
+ formula: frames = k * 2 ^ (value)
+
+ Substituting, we get: k = 0.00146
+
+ Further info on logarithmic nature:
+ frames to decay to sustain level 3 = 3 * frames to decay to
+ sustain level 9
+
+ Also no. of frames to 25% volume = roughly 1.85 * no. of frames to
+ 50% volume.
+
+ Frag it - just use linear approx.
+
+ ------------------------------------
+ Pete:
+ ms=((1<<value)*292)/10000
+ ------------------------------------
+
+
+3. Sustain rate value (linear mode)
+
+ Sustain rate range: 0 -> 127
+
+ Value | 48 | 52 | 56 | 60 | 64 | 68 | 72 |
+ -------------------------------------------
+ frames | 9 | 19 | 37 | 74 | 147| 293| 587|
+
+ Here, frames = no. of PAL frames for volume amplitude to go from 100%
+ to 0% (or vice-versa).
+
+ Same formula as for attack value, just a different value for k:
+
+ k = 0.00225
+
+ ie: frames = 0.00225 * 2 ^ (value / 4)
+
+ For emulation purposes:
+
+ %volume_increase_or_decrease_per_tick = 1 / frames
+
+ ------------------------------------
+ Pete:
+ ms=((1<<(value>>2))*450)/10000
+ ------------------------------------
+
+
+4. Release rate (linear mode)
+
+ Release rate range: 0 -> 31
+
+ Value | 13 | 14 | 15 | 16 | 17 |
+ ---------------------------------------------------------------
+ frames | 18 | 36 | 73 | 146| 292|
+
+ Here, frames = no. of PAL frames to decay from 100% vol to 0% vol
+ after "note-off" is triggered.
+
+ Formula: frames = k * 2 ^ (value)
+
+ And so: k = 0.00223
+
+ ------------------------------------
+ Pete:
+ ms=((1<<value)*446)/10000
+ ------------------------------------
+
+
+Other notes:
+
+Log stuff not figured out. You may get some clues from the "Decay rate"
+stuff above. For emu purposes it may not be important - use linear
+approx.
+
+To get timings in millisecs, multiply frames by 20.
+
+
+
+- James Higgs 17/6/2000
+james7780@yahoo.com
+
+//---------------------------------------------------------------
+
+OLD adsr mixing according to james' rules... has to be called
+every one millisecond
+
+
+ long v,v2,lT,l1,l2,l3;
+
+ if(s_chan[ch].bStop) // psx wants to stop? -> release phase
+ {
+ if(s_chan[ch].ADSR.ReleaseVal!=0) // -> release not 0: do release (if 0: stop right now)
+ {
+ if(!s_chan[ch].ADSR.ReleaseVol) // --> release just started? set up the release stuff
+ {
+ s_chan[ch].ADSR.ReleaseStartTime=s_chan[ch].ADSR.lTime;
+ s_chan[ch].ADSR.ReleaseVol=s_chan[ch].ADSR.lVolume;
+ s_chan[ch].ADSR.ReleaseTime = // --> calc how long does it take to reach the wanted sus level
+ (s_chan[ch].ADSR.ReleaseTime*
+ s_chan[ch].ADSR.ReleaseVol)/1024;
+ }
+ // -> NO release exp mode used (yet)
+ v=s_chan[ch].ADSR.ReleaseVol; // -> get last volume
+ lT=s_chan[ch].ADSR.lTime- // -> how much time is past?
+ s_chan[ch].ADSR.ReleaseStartTime;
+ l1=s_chan[ch].ADSR.ReleaseTime;
+
+ if(lT<l1) // -> we still have to release
+ {
+ v=v-((v*lT)/l1); // --> calc new volume
+ }
+ else // -> release is over: now really stop that sample
+ {v=0;s_chan[ch].bOn=0;s_chan[ch].ADSR.ReleaseVol=0;s_chan[ch].bNoise=0;}
+ }
+ else // -> release IS 0: release at once
+ {
+ v=0;s_chan[ch].bOn=0;s_chan[ch].ADSR.ReleaseVol=0;s_chan[ch].bNoise=0;
+ }
+ }
+ else
+ {//--------------------------------------------------// not in release phase:
+ v=1024;
+ lT=s_chan[ch].ADSR.lTime;
+ l1=s_chan[ch].ADSR.AttackTime;
+
+ if(lT<l1) // attack
+ { // no exp mode used (yet)
+// if(s_chan[ch].ADSR.AttackModeExp)
+// {
+// v=(v*lT)/l1;
+// }
+// else
+ {
+ v=(v*lT)/l1;
+ }
+ if(v==0) v=1;
+ }
+ else // decay
+ { // should be exp, but who cares? ;)
+ l2=s_chan[ch].ADSR.DecayTime;
+ v2=s_chan[ch].ADSR.SustainLevel;
+
+ lT-=l1;
+ if(lT<l2)
+ {
+ v-=(((v-v2)*lT)/l2);
+ }
+ else // sustain
+ { // no exp mode used (yet)
+ l3=s_chan[ch].ADSR.SustainTime;
+ lT-=l2;
+ if(s_chan[ch].ADSR.SustainModeDec>0)
+ {
+ if(l3!=0) v2+=((v-v2)*lT)/l3;
+ else v2=v;
+ }
+ else
+ {
+ if(l3!=0) v2-=(v2*lT)/l3;
+ else v2=v;
+ }
+
+ if(v2>v) v2=v;
+ if(v2<=0) {v2=0;s_chan[ch].bOn=0;s_chan[ch].ADSR.ReleaseVol=0;s_chan[ch].bNoise=0;}
+
+ v=v2;
+ }
+ }
+ }
+
+ //----------------------------------------------------//
+ // ok, done for this channel, so increase time
+
+ s_chan[ch].ADSR.lTime+=1; // 1 = 1.020408f ms;
+
+ if(v>1024) v=1024; // adjust volume
+ if(v<0) v=0;
+ s_chan[ch].ADSR.lVolume=v; // store act volume
+
+ return v; // return the volume factor
+*/
+
+
+//-----------------------------------------------------------------------------
+//-----------------------------------------------------------------------------
+//-----------------------------------------------------------------------------
+
+
+/*
+-----------------------------------------------------------------------------
+Neill Corlett
+Playstation SPU envelope timing notes
+-----------------------------------------------------------------------------
+
+This is preliminary. This may be wrong. But the model described herein fits
+all of my experimental data, and it's just simple enough to sound right.
+
+ADSR envelope level ranges from 0x00000000 to 0x7FFFFFFF internally.
+The value returned by channel reg 0xC is (envelope_level>>16).
+
+Each sample, an increment or decrement value will be added to or
+subtracted from this envelope level.
+
+Create the rate log table. The values double every 4 entries.
+ entry #0 = 4
+
+ 4, 5, 6, 7,
+ 8,10,12,14,
+ 16,20,24,28, ...
+
+ entry #40 = 4096...
+ entry #44 = 8192...
+ entry #48 = 16384...
+ entry #52 = 32768...
+ entry #56 = 65536...
+
+increments and decrements are in terms of ratelogtable[n]
+n may exceed the table bounds (plan on n being between -32 and 127).
+table values are all clipped between 0x00000000 and 0x3FFFFFFF
+
+when you "voice on", the envelope is always fully reset.
+(yes, it may click. the real thing does this too.)
+
+envelope level begins at zero.
+
+each state happens for at least 1 cycle
+(transitions are not instantaneous)
+this may result in some oddness: if the decay rate is uberfast, it will cut
+the envelope from full down to half in one sample, potentially skipping over
+the sustain level
+
+ATTACK
+------
+- if the envelope level has overflowed past the max, clip to 0x7FFFFFFF and
+ proceed to DECAY.
+
+Linear attack mode:
+- line extends upward to 0x7FFFFFFF
+- increment per sample is ratelogtable[(Ar^0x7F)-0x10]
+
+Logarithmic attack mode:
+if envelope_level < 0x60000000:
+ - line extends upward to 0x60000000
+ - increment per sample is ratelogtable[(Ar^0x7F)-0x10]
+else:
+ - line extends upward to 0x7FFFFFFF
+ - increment per sample is ratelogtable[(Ar^0x7F)-0x18]
+
+DECAY
+-----
+- if ((envelope_level>>27)&0xF) <= Sl, proceed to SUSTAIN.
+ Do not clip to the sustain level.
+- current line ends at (envelope_level & 0x07FFFFFF)
+- decrement per sample depends on (envelope_level>>28)&0x7
+ 0: ratelogtable[(4*(Dr^0x1F))-0x18+0]
+ 1: ratelogtable[(4*(Dr^0x1F))-0x18+4]
+ 2: ratelogtable[(4*(Dr^0x1F))-0x18+6]
+ 3: ratelogtable[(4*(Dr^0x1F))-0x18+8]
+ 4: ratelogtable[(4*(Dr^0x1F))-0x18+9]
+ 5: ratelogtable[(4*(Dr^0x1F))-0x18+10]
+ 6: ratelogtable[(4*(Dr^0x1F))-0x18+11]
+ 7: ratelogtable[(4*(Dr^0x1F))-0x18+12]
+ (note that this is the same as the release rate formula, except that
+ decay rates 10-1F aren't possible... those would be slower in theory)
+
+SUSTAIN
+-------
+- no terminating condition except for voice off
+- Sd=0 (increase) behavior is identical to ATTACK for both log and linear.
+- Sd=1 (decrease) behavior:
+Linear sustain decrease:
+- line extends to 0x00000000
+- decrement per sample is ratelogtable[(Sr^0x7F)-0x0F]
+Logarithmic sustain decrease:
+- current line ends at (envelope_level & 0x07FFFFFF)
+- decrement per sample depends on (envelope_level>>28)&0x7
+ 0: ratelogtable[(Sr^0x7F)-0x1B+0]
+ 1: ratelogtable[(Sr^0x7F)-0x1B+4]
+ 2: ratelogtable[(Sr^0x7F)-0x1B+6]
+ 3: ratelogtable[(Sr^0x7F)-0x1B+8]
+ 4: ratelogtable[(Sr^0x7F)-0x1B+9]
+ 5: ratelogtable[(Sr^0x7F)-0x1B+10]
+ 6: ratelogtable[(Sr^0x7F)-0x1B+11]
+ 7: ratelogtable[(Sr^0x7F)-0x1B+12]
+
+RELEASE
+-------
+- if the envelope level has overflowed to negative, clip to 0 and QUIT.
+
+Linear release mode:
+- line extends to 0x00000000
+- decrement per sample is ratelogtable[(4*(Rr^0x1F))-0x0C]
+
+Logarithmic release mode:
+- line extends to (envelope_level & 0x0FFFFFFF)
+- decrement per sample depends on (envelope_level>>28)&0x7
+ 0: ratelogtable[(4*(Rr^0x1F))-0x18+0]
+ 1: ratelogtable[(4*(Rr^0x1F))-0x18+4]
+ 2: ratelogtable[(4*(Rr^0x1F))-0x18+6]
+ 3: ratelogtable[(4*(Rr^0x1F))-0x18+8]
+ 4: ratelogtable[(4*(Rr^0x1F))-0x18+9]
+ 5: ratelogtable[(4*(Rr^0x1F))-0x18+10]
+ 6: ratelogtable[(4*(Rr^0x1F))-0x18+11]
+ 7: ratelogtable[(4*(Rr^0x1F))-0x18+12]
+
+-----------------------------------------------------------------------------
+*/
+
diff --git a/plugins/dfsound/adsr.h b/plugins/dfsound/adsr.h new file mode 100644 index 0000000..ff2af1f --- /dev/null +++ b/plugins/dfsound/adsr.h @@ -0,0 +1,19 @@ +/***************************************************************************
+ adsr.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+INLINE void StartADSR(int ch);
+INLINE int MixADSR(int ch);
diff --git a/plugins/dfsound/alsa.c b/plugins/dfsound/alsa.c new file mode 100644 index 0000000..2eba878 --- /dev/null +++ b/plugins/dfsound/alsa.c @@ -0,0 +1,158 @@ +/*************************************************************************** + alsa.c - description + ------------------- + begin : Sat Mar 01 2003 + copyright : (C) 2002 by Pete Bernert + email : BlackDove@addcom.de + ***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +#include "stdafx.h" + +#define _IN_OSS + +#include "externals.h" + +#define ALSA_PCM_NEW_HW_PARAMS_API +#define ALSA_PCM_NEW_SW_PARAMS_API +#include <alsa/asoundlib.h> + +static snd_pcm_t *handle = NULL; +static snd_pcm_uframes_t buffer_size; + +// SETUP SOUND +void SetupSound(void) +{ + snd_pcm_hw_params_t *hwparams; + snd_pcm_status_t *status; + unsigned int pspeed; + int pchannels; + int format; + unsigned int buffer_time = 100000; + unsigned int period_time = buffer_time / 4; + int err; + + if (iDisStereo) pchannels = 1; + else pchannels=2; + + pspeed = 44100; + format = SND_PCM_FORMAT_S16; + + if ((err = snd_pcm_open(&handle, "default", + SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) + { + printf("Audio open error: %s\n", snd_strerror(err)); + return; + } + + if((err = snd_pcm_nonblock(handle, 0))<0) + { + printf("Can't set blocking moded: %s\n", snd_strerror(err)); + return; + } + + snd_pcm_hw_params_alloca(&hwparams); + + if((err=snd_pcm_hw_params_any(handle, hwparams))<0) + { + printf("Broken configuration for this PCM: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED))<0) + { + printf("Access type not available: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params_set_format(handle, hwparams, format))<0) + { + printf("Sample format not available: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params_set_channels(handle, hwparams, pchannels))<0) + { + printf("Channels count not available: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params_set_rate_near(handle, hwparams, &pspeed, 0))<0) + { + printf("Rate not available: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0))<0) + { + printf("Buffer time error: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0))<0) + { + printf("Period time error: %s\n", snd_strerror(err)); + return; + } + + if((err=snd_pcm_hw_params(handle, hwparams))<0) + { + printf("Unable to install hw params: %s\n", snd_strerror(err)); + return; + } + + snd_pcm_status_alloca(&status); + if((err=snd_pcm_status(handle, status))<0) + { + printf("Unable to get status: %s\n", snd_strerror(err)); + return; + } + + buffer_size = snd_pcm_status_get_avail(status); +} + +// REMOVE SOUND +void RemoveSound(void) +{ + if(handle != NULL) + { + snd_pcm_drop(handle); + snd_pcm_close(handle); + handle = NULL; + } +} + +// GET BYTES BUFFERED +unsigned long SoundGetBytesBuffered(void) +{ + unsigned long l; + + if (handle == NULL) // failed to open? + return SOUNDSIZE; + l = snd_pcm_avail(handle); + if (l < 0) return 0; + if (l < buffer_size / 2) // can we write in at least the half of fragments? + l = SOUNDSIZE; // -> no? wait + else l = 0; // -> else go on + + return l; +} + +// FEED SOUND DATA +void SoundFeedStreamData(unsigned char* pSound,long lBytes) +{ + if (handle == NULL) return; + + if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN) + snd_pcm_prepare(handle); + snd_pcm_writei(handle,pSound, + iDisStereo ? lBytes / 2 : lBytes / 4); +} diff --git a/plugins/dfsound/cfg.c b/plugins/dfsound/cfg.c new file mode 100644 index 0000000..2acb9c3 --- /dev/null +++ b/plugins/dfsound/cfg.c @@ -0,0 +1,167 @@ +/***************************************************************************
+ cfg.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_CFG
+
+#include "externals.h"
+
+////////////////////////////////////////////////////////////////////////
+// LINUX CONFIG/ABOUT HANDLING
+////////////////////////////////////////////////////////////////////////
+
+#include <unistd.h>
+
+////////////////////////////////////////////////////////////////////////
+// START EXTERNAL CFG TOOL
+////////////////////////////////////////////////////////////////////////
+
+void StartCfgTool(char * pCmdLine)
+{
+ FILE * cf;
+ char filename[255];
+
+ strcpy(filename,"cfgDFSound");
+ cf=fopen(filename,"rb");
+ if(cf!=NULL)
+ {
+ fclose(cf);
+ if(fork()==0)
+ {
+ execl("./cfgDFSound","cfgDFSound",pCmdLine,NULL);
+ exit(0);
+ }
+ }
+ else
+ {
+ strcpy(filename,"cfg/cfgDFSound");
+ cf=fopen(filename,"rb");
+ if(cf!=NULL)
+ {
+ fclose(cf);
+ if(fork()==0)
+ {
+ chdir("cfg");
+ execl("./cfgDFSound","cfgDFSound",pCmdLine,NULL);
+ exit(0);
+ }
+ }
+ else
+ {
+ sprintf(filename,"%s/cfgDFSound",getenv("HOME"));
+ cf=fopen(filename,"rb");
+ if(cf!=NULL)
+ {
+ fclose(cf);
+ if(fork()==0)
+ {
+ chdir(getenv("HOME"));
+ execl("./cfgDFSound","cfgDFSound",pCmdLine,NULL);
+ exit(0);
+ }
+ }
+ else printf("Sound error: cfgDFSound not found!\n");
+ }
+ }
+}
+
+/////////////////////////////////////////////////////////
+// READ LINUX CONFIG FILE
+/////////////////////////////////////////////////////////
+
+void ReadConfigFile(void)
+{
+ FILE *in;char t[256];int len;
+ char * pB, * p;
+
+ strcpy(t,"dfsound.cfg");
+ in = fopen(t,"rb");
+ if(!in)
+ {
+ strcpy(t,"cfg/dfsound.cfg");
+ in = fopen(t,"rb");
+ if(!in)
+ {
+ sprintf(t,"%s/dfsound.cfg",getenv("HOME"));
+ in = fopen(t,"rb");
+ if(!in) return;
+ }
+ }
+
+ pB = (char *)malloc(32767);
+ memset(pB,0,32767);
+
+ len = fread(pB, 1, 32767, in);
+ fclose(in);
+
+ strcpy(t,"\nVolume");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iVolume=4-atoi(p+len);
+ if(iVolume<1) iVolume=1;
+ if(iVolume>4) iVolume=4;
+
+ strcpy(t,"\nXAPitch");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iXAPitch=atoi(p+len);
+ if(iXAPitch<0) iXAPitch=0;
+ if(iXAPitch>1) iXAPitch=1;
+
+ strcpy(t,"\nHighCompMode");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iUseTimer=atoi(p+len);
+ if(iUseTimer<0) iUseTimer=0;
+ // note: timer mode 1 (win time events) is not supported
+ // in linux. But timer mode 2 (spuupdate) is safe to use.
+ if(iUseTimer) iUseTimer=2;
+
+ strcpy(t,"\nSPUIRQWait");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iSPUIRQWait=atoi(p+len);
+ if(iSPUIRQWait<0) iSPUIRQWait=0;
+ if(iSPUIRQWait>1) iSPUIRQWait=1;
+
+ strcpy(t,"\nUseReverb");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iUseReverb=atoi(p+len);
+ if(iUseReverb<0) iUseReverb=0;
+ if(iUseReverb>2) iUseReverb=2;
+
+ strcpy(t,"\nUseInterpolation");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iUseInterpolation=atoi(p+len);
+ if(iUseInterpolation<0) iUseInterpolation=0;
+ if(iUseInterpolation>3) iUseInterpolation=3;
+
+ strcpy(t,"\nDisStereo");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;}
+ if(p) iDisStereo=atoi(p+len);
+ if(iDisStereo<0) iDisStereo=0;
+ if(iDisStereo>1) iDisStereo=1;
+
+ free(pB);
+}
+
+/////////////////////////////////////////////////////////
+// READ CONFIG called by spu funcs
+/////////////////////////////////////////////////////////
+
+void ReadConfig(void)
+{
+ iVolume=2;
+ iXAPitch=0;
+ iSPUIRQWait=1;
+ iUseTimer=2;
+ iUseReverb=2;
+ iUseInterpolation=2;
+ iDisStereo=0;
+
+ ReadConfigFile();
+}
diff --git a/plugins/dfsound/cfg.h b/plugins/dfsound/cfg.h new file mode 100644 index 0000000..f64d6d6 --- /dev/null +++ b/plugins/dfsound/cfg.h @@ -0,0 +1,19 @@ +/***************************************************************************
+ cfg.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+void ReadConfig(void);
+void StartCfgTool(char * pCmdLine);
diff --git a/plugins/dfsound/dma.c b/plugins/dfsound/dma.c new file mode 100644 index 0000000..f92d066 --- /dev/null +++ b/plugins/dfsound/dma.c @@ -0,0 +1,97 @@ +/***************************************************************************
+ dma.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_DMA
+
+#include "externals.h"
+
+////////////////////////////////////////////////////////////////////////
+// READ DMA (one value)
+////////////////////////////////////////////////////////////////////////
+
+unsigned short CALLBACK SPUreadDMA(void)
+{
+ unsigned short s=spuMem[spuAddr>>1];
+ spuAddr+=2;
+ if(spuAddr>0x7ffff) spuAddr=0;
+
+ iSpuAsyncWait=0;
+
+ return s;
+}
+
+////////////////////////////////////////////////////////////////////////
+// READ DMA (many values)
+////////////////////////////////////////////////////////////////////////
+
+void CALLBACK SPUreadDMAMem(unsigned short * pusPSXMem,int iSize)
+{
+ int i;
+
+ for(i=0;i<iSize;i++)
+ {
+ *pusPSXMem++=spuMem[spuAddr>>1]; // spu addr got by writeregister
+ spuAddr+=2; // inc spu addr
+ if(spuAddr>0x7ffff) spuAddr=0; // wrap
+ }
+
+ iSpuAsyncWait=0;
+}
+
+////////////////////////////////////////////////////////////////////////
+////////////////////////////////////////////////////////////////////////
+////////////////////////////////////////////////////////////////////////
+
+// to investigate: do sound data updates by writedma affect spu
+// irqs? Will an irq be triggered, if new data is written to
+// the memory irq address?
+
+////////////////////////////////////////////////////////////////////////
+// WRITE DMA (one value)
+////////////////////////////////////////////////////////////////////////
+
+void CALLBACK SPUwriteDMA(unsigned short val)
+{
+ spuMem[spuAddr>>1] = val; // spu addr got by writeregister
+
+ spuAddr+=2; // inc spu addr
+ if(spuAddr>0x7ffff) spuAddr=0; // wrap
+
+ iSpuAsyncWait=0;
+}
+
+////////////////////////////////////////////////////////////////////////
+// WRITE DMA (many values)
+////////////////////////////////////////////////////////////////////////
+
+void CALLBACK SPUwriteDMAMem(unsigned short * pusPSXMem,int iSize)
+{
+ int i;
+
+ for(i=0;i<iSize;i++)
+ {
+ spuMem[spuAddr>>1] = *pusPSXMem++; // spu addr got by writeregister
+ spuAddr+=2; // inc spu addr
+ if(spuAddr>0x7ffff) spuAddr=0; // wrap
+ }
+
+ iSpuAsyncWait=0;
+}
+
+////////////////////////////////////////////////////////////////////////
diff --git a/plugins/dfsound/dma.h b/plugins/dfsound/dma.h new file mode 100644 index 0000000..440536f --- /dev/null +++ b/plugins/dfsound/dma.h @@ -0,0 +1,31 @@ +/***************************************************************************
+ dma.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+//*************************************************************************//
+// History of changes:
+//
+// 2002/05/15 - Pete
+// - generic cleanup for the Peops release
+//
+//*************************************************************************//
+
+
+unsigned short CALLBACK SPUreadDMA(void);
+void CALLBACK SPUreadDMAMem(unsigned short * pusPSXMem,int iSize);
+void CALLBACK SPUwriteDMA(unsigned short val);
+void CALLBACK SPUwriteDMAMem(unsigned short * pusPSXMem,int iSize);
diff --git a/plugins/dfsound/dsoundoss.h b/plugins/dfsound/dsoundoss.h new file mode 100644 index 0000000..3702312 --- /dev/null +++ b/plugins/dfsound/dsoundoss.h @@ -0,0 +1,22 @@ +/*************************************************************************** + dsoundoss.h - description + ------------------- + begin : Wed May 15 2002 + copyright : (C) 2002 by Pete Bernert + email : BlackDove@addcom.de + ***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +void SetupSound(void); +void RemoveSound(void); +unsigned long SoundGetBytesBuffered(void); +void SoundFeedStreamData(unsigned char* pSound,long lBytes); +unsigned long timeGetTime_spu(); diff --git a/plugins/dfsound/externals.h b/plugins/dfsound/externals.h new file mode 100644 index 0000000..f856204 --- /dev/null +++ b/plugins/dfsound/externals.h @@ -0,0 +1,286 @@ +/***************************************************************************
+ externals.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include <stdint.h>
+
+/////////////////////////////////////////////////////////
+// generic defines
+/////////////////////////////////////////////////////////
+
+#define PSE_LT_SPU 4
+#define PSE_SPU_ERR_SUCCESS 0
+#define PSE_SPU_ERR -60
+#define PSE_SPU_ERR_NOTCONFIGURED PSE_SPU_ERR - 1
+#define PSE_SPU_ERR_INIT PSE_SPU_ERR - 2
+#ifndef max
+#define max(a,b) (((a) > (b)) ? (a) : (b))
+#define min(a,b) (((a) < (b)) ? (a) : (b))
+#endif
+
+////////////////////////////////////////////////////////////////////////
+// spu defines
+////////////////////////////////////////////////////////////////////////
+
+// sound buffer sizes
+// 400 ms complete sound buffer
+#define SOUNDSIZE 70560
+// 137 ms test buffer... if less than that is buffered, a new upload will happen
+#define TESTSIZE 24192
+
+// num of channels
+#define MAXCHAN 24
+
+// ~ 1 ms of data
+#define NSSIZE 45
+
+///////////////////////////////////////////////////////////
+// struct defines
+///////////////////////////////////////////////////////////
+
+// ADSR INFOS PER CHANNEL
+typedef struct
+{
+ int AttackModeExp;
+ long AttackTime;
+ long DecayTime;
+ long SustainLevel;
+ int SustainModeExp;
+ long SustainModeDec;
+ long SustainTime;
+ int ReleaseModeExp;
+ unsigned long ReleaseVal;
+ long ReleaseTime;
+ long ReleaseStartTime;
+ long ReleaseVol;
+ long lTime;
+ long lVolume;
+} ADSRInfo;
+
+typedef struct
+{
+ int State;
+ int AttackModeExp;
+ int AttackRate;
+ int DecayRate;
+ int SustainLevel;
+ int SustainModeExp;
+ int SustainIncrease;
+ int SustainRate;
+ int ReleaseModeExp;
+ int ReleaseRate;
+ int EnvelopeVol;
+ long lVolume;
+ long lDummy1;
+ long lDummy2;
+} ADSRInfoEx;
+
+///////////////////////////////////////////////////////////
+
+// Tmp Flags
+
+// used for debug channel muting
+#define FLAG_MUTE 1
+
+// used for simple interpolation
+#define FLAG_IPOL0 2
+#define FLAG_IPOL1 4
+
+///////////////////////////////////////////////////////////
+
+// MAIN CHANNEL STRUCT
+typedef struct
+{
+ // no mutexes used anymore... don't need them to sync access
+ //HANDLE hMutex;
+
+ int bNew; // start flag
+
+ int iSBPos; // mixing stuff
+ int spos;
+ int sinc;
+ int SB[32+32]; // Pete added another 32 dwords in 1.6 ... prevents overflow issues with gaussian/cubic interpolation (thanx xodnizel!), and can be used for even better interpolations, eh? :)
+ int sval;
+
+ unsigned char * pStart; // start ptr into sound mem
+ unsigned char * pCurr; // current pos in sound mem
+ unsigned char * pLoop; // loop ptr in sound mem
+
+ int bOn; // is channel active (sample playing?)
+ int bStop; // is channel stopped (sample _can_ still be playing, ADSR Release phase)
+ int bReverb; // can we do reverb on this channel? must have ctrl register bit, to get active
+ int iActFreq; // current psx pitch
+ int iUsedFreq; // current pc pitch
+ int iLeftVolume; // left volume
+ int iLeftVolRaw; // left psx volume value
+ int bIgnoreLoop; // ignore loop bit, if an external loop address is used
+ int iMute; // mute mode
+ int iRightVolume; // right volume
+ int iRightVolRaw; // right psx volume value
+ int iRawPitch; // raw pitch (0...3fff)
+ int iIrqDone; // debug irq done flag
+ int s_1; // last decoding infos
+ int s_2;
+ int bRVBActive; // reverb active flag
+ int iRVBOffset; // reverb offset
+ int iRVBRepeat; // reverb repeat
+ int bNoise; // noise active flag
+ int bFMod; // freq mod (0=off, 1=sound channel, 2=freq channel)
+ int iRVBNum; // another reverb helper
+ int iOldNoise; // old noise val for this channel
+ ADSRInfo ADSR; // active ADSR settings
+ ADSRInfoEx ADSRX; // next ADSR settings (will be moved to active on sample start)
+} SPUCHAN;
+
+///////////////////////////////////////////////////////////
+
+typedef struct
+{
+ int StartAddr; // reverb area start addr in samples
+ int CurrAddr; // reverb area curr addr in samples
+
+ int VolLeft;
+ int VolRight;
+ int iLastRVBLeft;
+ int iLastRVBRight;
+ int iRVBLeft;
+ int iRVBRight;
+
+ int FB_SRC_A; // (offset)
+ int FB_SRC_B; // (offset)
+ int IIR_ALPHA; // (coef.)
+ int ACC_COEF_A; // (coef.)
+ int ACC_COEF_B; // (coef.)
+ int ACC_COEF_C; // (coef.)
+ int ACC_COEF_D; // (coef.)
+ int IIR_COEF; // (coef.)
+ int FB_ALPHA; // (coef.)
+ int FB_X; // (coef.)
+ int IIR_DEST_A0; // (offset)
+ int IIR_DEST_A1; // (offset)
+ int ACC_SRC_A0; // (offset)
+ int ACC_SRC_A1; // (offset)
+ int ACC_SRC_B0; // (offset)
+ int ACC_SRC_B1; // (offset)
+ int IIR_SRC_A0; // (offset)
+ int IIR_SRC_A1; // (offset)
+ int IIR_DEST_B0; // (offset)
+ int IIR_DEST_B1; // (offset)
+ int ACC_SRC_C0; // (offset)
+ int ACC_SRC_C1; // (offset)
+ int ACC_SRC_D0; // (offset)
+ int ACC_SRC_D1; // (offset)
+ int IIR_SRC_B1; // (offset)
+ int IIR_SRC_B0; // (offset)
+ int MIX_DEST_A0; // (offset)
+ int MIX_DEST_A1; // (offset)
+ int MIX_DEST_B0; // (offset)
+ int MIX_DEST_B1; // (offset)
+ int IN_COEF_L; // (coef.)
+ int IN_COEF_R; // (coef.)
+} REVERBInfo;
+
+///////////////////////////////////////////////////////////
+// SPU.C globals
+///////////////////////////////////////////////////////////
+
+#ifndef _IN_SPU
+
+// psx buffers / addresses
+
+extern unsigned short regArea[];
+extern unsigned short spuMem[];
+extern unsigned char * spuMemC;
+extern unsigned char * pSpuIrq;
+extern unsigned char * pSpuBuffer;
+
+// user settings
+
+extern int iVolume;
+extern int iXAPitch;
+extern int iUseTimer;
+extern int iSPUIRQWait;
+extern int iDebugMode;
+extern int iRecordMode;
+extern int iUseReverb;
+extern int iUseInterpolation;
+extern int iDisStereo;
+// MISC
+
+extern int iSpuAsyncWait;
+
+extern SPUCHAN s_chan[];
+extern REVERBInfo rvb;
+
+extern unsigned long dwNoiseVal;
+extern unsigned short spuCtrl;
+extern unsigned short spuStat;
+extern unsigned short spuIrq;
+extern unsigned long spuAddr;
+extern int bEndThread;
+extern int bThreadEnded;
+extern int bSpuInit;
+extern unsigned long dwNewChannel;
+
+extern int SSumR[];
+extern int SSumL[];
+extern int iCycle;
+extern short * pS;
+
+extern void (CALLBACK *cddavCallback)(unsigned short,unsigned short);
+
+#endif
+
+///////////////////////////////////////////////////////////
+// XA.C globals
+///////////////////////////////////////////////////////////
+
+#ifndef _IN_XA
+
+extern xa_decode_t * xapGlobal;
+
+extern uint32_t * XAFeed;
+extern uint32_t * XAPlay;
+extern uint32_t * XAStart;
+extern uint32_t * XAEnd;
+
+extern uint32_t XARepeat;
+extern uint32_t XALastVal;
+
+extern uint32_t * CDDAFeed;
+extern uint32_t * CDDAPlay;
+extern uint32_t * CDDAStart;
+extern uint32_t * CDDAEnd;
+
+extern int iLeftXAVol;
+extern int iRightXAVol;
+
+#endif
+
+///////////////////////////////////////////////////////////
+// REVERB.C globals
+///////////////////////////////////////////////////////////
+
+#ifndef _IN_REVERB
+
+extern int * sRVBPlay;
+extern int * sRVBEnd;
+extern int * sRVBStart;
+extern int iReverbOff;
+extern int iReverbRepeat;
+extern int iReverbNum;
+
+#endif
diff --git a/plugins/dfsound/freeze.c b/plugins/dfsound/freeze.c new file mode 100644 index 0000000..12fdc1f --- /dev/null +++ b/plugins/dfsound/freeze.c @@ -0,0 +1,214 @@ +/***************************************************************************
+ freeze.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_FREEZE
+
+#include "externals.h"
+#include "registers.h"
+#include "spu.h"
+#include "regs.h"
+
+////////////////////////////////////////////////////////////////////////
+// freeze structs
+////////////////////////////////////////////////////////////////////////
+
+typedef struct
+{
+ char szSPUName[8];
+ uint32_t ulFreezeVersion;
+ uint32_t ulFreezeSize;
+ unsigned char cSPUPort[0x200];
+ unsigned char cSPURam[0x80000];
+ xa_decode_t xaS;
+} SPUFreeze_t;
+
+typedef struct
+{
+ unsigned short spuIrq;
+ uint32_t pSpuIrq;
+ uint32_t spuAddr;
+ uint32_t dummy1;
+ uint32_t dummy2;
+ uint32_t dummy3;
+
+ SPUCHAN s_chan[MAXCHAN];
+
+} SPUOSSFreeze_t;
+
+////////////////////////////////////////////////////////////////////////
+
+void LoadStateV5(SPUFreeze_t * pF); // newest version
+void LoadStateUnknown(SPUFreeze_t * pF); // unknown format
+
+extern int lastch;
+
+////////////////////////////////////////////////////////////////////////
+// SPUFREEZE: called by main emu on savestate load/save
+////////////////////////////////////////////////////////////////////////
+
+long CALLBACK SPUfreeze(uint32_t ulFreezeMode,SPUFreeze_t * pF)
+{
+ int i;SPUOSSFreeze_t * pFO;
+
+ if(!pF) return 0; // first check
+
+ if(ulFreezeMode) // info or save?
+ {//--------------------------------------------------//
+ if(ulFreezeMode==1)
+ memset(pF,0,sizeof(SPUFreeze_t)+sizeof(SPUOSSFreeze_t));
+
+ strcpy(pF->szSPUName,"PBOSS");
+ pF->ulFreezeVersion=5;
+ pF->ulFreezeSize=sizeof(SPUFreeze_t)+sizeof(SPUOSSFreeze_t);
+
+ if(ulFreezeMode==2) return 1; // info mode? ok, bye
+ // save mode:
+ RemoveTimer(); // stop timer
+
+ memcpy(pF->cSPURam,spuMem,0x80000); // copy common infos
+ memcpy(pF->cSPUPort,regArea,0x200);
+
+ if(xapGlobal && XAPlay!=XAFeed) // some xa
+ {
+ pF->xaS=*xapGlobal;
+ }
+ else
+ memset(&pF->xaS,0,sizeof(xa_decode_t)); // or clean xa
+
+ pFO=(SPUOSSFreeze_t *)(pF+1); // store special stuff
+
+ pFO->spuIrq=spuIrq;
+ if(pSpuIrq) pFO->pSpuIrq = (unsigned long)pSpuIrq-(unsigned long)spuMemC;
+
+ pFO->spuAddr=spuAddr;
+ if(pFO->spuAddr==0) pFO->spuAddr=0xbaadf00d;
+
+ for(i=0;i<MAXCHAN;i++)
+ {
+ memcpy((void *)&pFO->s_chan[i],(void *)&s_chan[i],sizeof(SPUCHAN));
+ if(pFO->s_chan[i].pStart)
+ pFO->s_chan[i].pStart-=(unsigned long)spuMemC;
+ if(pFO->s_chan[i].pCurr)
+ pFO->s_chan[i].pCurr-=(unsigned long)spuMemC;
+ if(pFO->s_chan[i].pLoop)
+ pFO->s_chan[i].pLoop-=(unsigned long)spuMemC;
+ }
+
+ SetupTimer(); // sound processing on again
+
+ return 1;
+ //--------------------------------------------------//
+ }
+
+ if(ulFreezeMode!=0) return 0; // bad mode? bye
+
+ RemoveTimer(); // we stop processing while doing the save!
+
+ memcpy(spuMem,pF->cSPURam,0x80000); // get ram
+ memcpy(regArea,pF->cSPUPort,0x200);
+
+ if(pF->xaS.nsamples<=4032) // start xa again
+ SPUplayADPCMchannel(&pF->xaS);
+
+ xapGlobal=0;
+
+ if(!strcmp(pF->szSPUName,"PBOSS") && pF->ulFreezeVersion==5)
+ LoadStateV5(pF);
+ else LoadStateUnknown(pF);
+
+ lastch = -1;
+
+ // repair some globals
+ for(i=0;i<=62;i+=2)
+ SPUwriteRegister(H_Reverb+i,regArea[(H_Reverb+i-0xc00)>>1]);
+ SPUwriteRegister(H_SPUReverbAddr,regArea[(H_SPUReverbAddr-0xc00)>>1]);
+ SPUwriteRegister(H_SPUrvolL,regArea[(H_SPUrvolL-0xc00)>>1]);
+ SPUwriteRegister(H_SPUrvolR,regArea[(H_SPUrvolR-0xc00)>>1]);
+
+ SPUwriteRegister(H_SPUctrl,(unsigned short)(regArea[(H_SPUctrl-0xc00)>>1]|0x4000));
+ SPUwriteRegister(H_SPUstat,regArea[(H_SPUstat-0xc00)>>1]);
+ SPUwriteRegister(H_CDLeft,regArea[(H_CDLeft-0xc00)>>1]);
+ SPUwriteRegister(H_CDRight,regArea[(H_CDRight-0xc00)>>1]);
+
+ // fix to prevent new interpolations from crashing
+ for(i=0;i<MAXCHAN;i++) s_chan[i].SB[28]=0;
+
+ SetupTimer(); // start sound processing again
+
+ return 1;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+void LoadStateV5(SPUFreeze_t * pF)
+{
+ int i;SPUOSSFreeze_t * pFO;
+
+ pFO=(SPUOSSFreeze_t *)(pF+1);
+
+ spuIrq = pFO->spuIrq;
+ if(pFO->pSpuIrq) pSpuIrq = pFO->pSpuIrq+spuMemC; else pSpuIrq=NULL;
+
+ if(pFO->spuAddr)
+ {
+ spuAddr = pFO->spuAddr;
+ if (spuAddr == 0xbaadf00d) spuAddr = 0;
+ }
+
+ for(i=0;i<MAXCHAN;i++)
+ {
+ memcpy((void *)&s_chan[i],(void *)&pFO->s_chan[i],sizeof(SPUCHAN));
+
+ s_chan[i].pStart+=(unsigned long)spuMemC;
+ s_chan[i].pCurr+=(unsigned long)spuMemC;
+ s_chan[i].pLoop+=(unsigned long)spuMemC;
+ s_chan[i].iMute=0;
+ s_chan[i].iIrqDone=0;
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+void LoadStateUnknown(SPUFreeze_t * pF)
+{
+ int i;
+
+ for(i=0;i<MAXCHAN;i++)
+ {
+ s_chan[i].bOn=0;
+ s_chan[i].bNew=0;
+ s_chan[i].bStop=0;
+ s_chan[i].ADSR.lVolume=0;
+ s_chan[i].pLoop=spuMemC;
+ s_chan[i].pStart=spuMemC;
+ s_chan[i].pLoop=spuMemC;
+ s_chan[i].iMute=0;
+ s_chan[i].iIrqDone=0;
+ }
+
+ dwNewChannel=0;
+ pSpuIrq=0;
+
+ for(i=0;i<0xc0;i++)
+ {
+ SPUwriteRegister(0x1f801c00+i*2,regArea[i]);
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
diff --git a/plugins/dfsound/gauss_i.h b/plugins/dfsound/gauss_i.h new file mode 100644 index 0000000..5a3a676 --- /dev/null +++ b/plugins/dfsound/gauss_i.h @@ -0,0 +1,150 @@ +/***************************************************************************
+ gauss_i.h - description
+ -----------------------
+ begin : Sun Feb 08 2003
+ copyright : (C) 2003 by Chris Moeller, eh, whatever
+ email : chris@kode54.tk
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#ifndef GAUSS_H
+#define GAUSS_H
+
+const int gauss[]={
+ 0x172, 0x519, 0x176, 0x000, 0x16E, 0x519, 0x17A, 0x000,
+ 0x16A, 0x518, 0x17D, 0x000, 0x166, 0x518, 0x181, 0x000,
+ 0x162, 0x518, 0x185, 0x000, 0x15F, 0x518, 0x189, 0x000,
+ 0x15B, 0x518, 0x18D, 0x000, 0x157, 0x517, 0x191, 0x000,
+ 0x153, 0x517, 0x195, 0x000, 0x150, 0x517, 0x19A, 0x000,
+ 0x14C, 0x516, 0x19E, 0x000, 0x148, 0x516, 0x1A2, 0x000,
+ 0x145, 0x515, 0x1A6, 0x000, 0x141, 0x514, 0x1AA, 0x000,
+ 0x13E, 0x514, 0x1AE, 0x000, 0x13A, 0x513, 0x1B2, 0x000,
+ 0x137, 0x512, 0x1B7, 0x001, 0x133, 0x511, 0x1BB, 0x001,
+ 0x130, 0x511, 0x1BF, 0x001, 0x12C, 0x510, 0x1C3, 0x001,
+ 0x129, 0x50F, 0x1C8, 0x001, 0x125, 0x50E, 0x1CC, 0x001,
+ 0x122, 0x50D, 0x1D0, 0x001, 0x11E, 0x50C, 0x1D5, 0x001,
+ 0x11B, 0x50B, 0x1D9, 0x001, 0x118, 0x50A, 0x1DD, 0x001,
+ 0x114, 0x508, 0x1E2, 0x001, 0x111, 0x507, 0x1E6, 0x002,
+ 0x10E, 0x506, 0x1EB, 0x002, 0x10B, 0x504, 0x1EF, 0x002,
+ 0x107, 0x503, 0x1F3, 0x002, 0x104, 0x502, 0x1F8, 0x002,
+ 0x101, 0x500, 0x1FC, 0x002, 0x0FE, 0x4FF, 0x201, 0x002,
+ 0x0FB, 0x4FD, 0x205, 0x003, 0x0F8, 0x4FB, 0x20A, 0x003,
+ 0x0F5, 0x4FA, 0x20F, 0x003, 0x0F2, 0x4F8, 0x213, 0x003,
+ 0x0EF, 0x4F6, 0x218, 0x003, 0x0EC, 0x4F5, 0x21C, 0x004,
+ 0x0E9, 0x4F3, 0x221, 0x004, 0x0E6, 0x4F1, 0x226, 0x004,
+ 0x0E3, 0x4EF, 0x22A, 0x004, 0x0E0, 0x4ED, 0x22F, 0x004,
+ 0x0DD, 0x4EB, 0x233, 0x005, 0x0DA, 0x4E9, 0x238, 0x005,
+ 0x0D7, 0x4E7, 0x23D, 0x005, 0x0D4, 0x4E5, 0x241, 0x005,
+ 0x0D2, 0x4E3, 0x246, 0x006, 0x0CF, 0x4E0, 0x24B, 0x006,
+ 0x0CC, 0x4DE, 0x250, 0x006, 0x0C9, 0x4DC, 0x254, 0x006,
+ 0x0C7, 0x4D9, 0x259, 0x007, 0x0C4, 0x4D7, 0x25E, 0x007,
+ 0x0C1, 0x4D5, 0x263, 0x007, 0x0BF, 0x4D2, 0x267, 0x008,
+ 0x0BC, 0x4D0, 0x26C, 0x008, 0x0BA, 0x4CD, 0x271, 0x008,
+ 0x0B7, 0x4CB, 0x276, 0x009, 0x0B4, 0x4C8, 0x27B, 0x009,
+ 0x0B2, 0x4C5, 0x280, 0x009, 0x0AF, 0x4C3, 0x284, 0x00A,
+ 0x0AD, 0x4C0, 0x289, 0x00A, 0x0AB, 0x4BD, 0x28E, 0x00A,
+ 0x0A8, 0x4BA, 0x293, 0x00B, 0x0A6, 0x4B7, 0x298, 0x00B,
+ 0x0A3, 0x4B5, 0x29D, 0x00B, 0x0A1, 0x4B2, 0x2A2, 0x00C,
+ 0x09F, 0x4AF, 0x2A6, 0x00C, 0x09C, 0x4AC, 0x2AB, 0x00D,
+ 0x09A, 0x4A9, 0x2B0, 0x00D, 0x098, 0x4A6, 0x2B5, 0x00E,
+ 0x096, 0x4A2, 0x2BA, 0x00E, 0x093, 0x49F, 0x2BF, 0x00F,
+ 0x091, 0x49C, 0x2C4, 0x00F, 0x08F, 0x499, 0x2C9, 0x00F,
+ 0x08D, 0x496, 0x2CE, 0x010, 0x08B, 0x492, 0x2D3, 0x010,
+ 0x089, 0x48F, 0x2D8, 0x011, 0x086, 0x48C, 0x2DC, 0x011,
+ 0x084, 0x488, 0x2E1, 0x012, 0x082, 0x485, 0x2E6, 0x013,
+ 0x080, 0x481, 0x2EB, 0x013, 0x07E, 0x47E, 0x2F0, 0x014,
+ 0x07C, 0x47A, 0x2F5, 0x014, 0x07A, 0x477, 0x2FA, 0x015,
+ 0x078, 0x473, 0x2FF, 0x015, 0x076, 0x470, 0x304, 0x016,
+ 0x075, 0x46C, 0x309, 0x017, 0x073, 0x468, 0x30E, 0x017,
+ 0x071, 0x465, 0x313, 0x018, 0x06F, 0x461, 0x318, 0x018,
+ 0x06D, 0x45D, 0x31D, 0x019, 0x06B, 0x459, 0x322, 0x01A,
+ 0x06A, 0x455, 0x326, 0x01B, 0x068, 0x452, 0x32B, 0x01B,
+ 0x066, 0x44E, 0x330, 0x01C, 0x064, 0x44A, 0x335, 0x01D,
+ 0x063, 0x446, 0x33A, 0x01D, 0x061, 0x442, 0x33F, 0x01E,
+ 0x05F, 0x43E, 0x344, 0x01F, 0x05E, 0x43A, 0x349, 0x020,
+ 0x05C, 0x436, 0x34E, 0x020, 0x05A, 0x432, 0x353, 0x021,
+ 0x059, 0x42E, 0x357, 0x022, 0x057, 0x42A, 0x35C, 0x023,
+ 0x056, 0x425, 0x361, 0x024, 0x054, 0x421, 0x366, 0x024,
+ 0x053, 0x41D, 0x36B, 0x025, 0x051, 0x419, 0x370, 0x026,
+ 0x050, 0x415, 0x374, 0x027, 0x04E, 0x410, 0x379, 0x028,
+ 0x04D, 0x40C, 0x37E, 0x029, 0x04C, 0x408, 0x383, 0x02A,
+ 0x04A, 0x403, 0x388, 0x02B, 0x049, 0x3FF, 0x38C, 0x02C,
+ 0x047, 0x3FB, 0x391, 0x02D, 0x046, 0x3F6, 0x396, 0x02E,
+ 0x045, 0x3F2, 0x39B, 0x02F, 0x043, 0x3ED, 0x39F, 0x030,
+ 0x042, 0x3E9, 0x3A4, 0x031, 0x041, 0x3E5, 0x3A9, 0x032,
+ 0x040, 0x3E0, 0x3AD, 0x033, 0x03E, 0x3DC, 0x3B2, 0x034,
+ 0x03D, 0x3D7, 0x3B7, 0x035, 0x03C, 0x3D2, 0x3BB, 0x036,
+ 0x03B, 0x3CE, 0x3C0, 0x037, 0x03A, 0x3C9, 0x3C5, 0x038,
+ 0x038, 0x3C5, 0x3C9, 0x03A, 0x037, 0x3C0, 0x3CE, 0x03B,
+ 0x036, 0x3BB, 0x3D2, 0x03C, 0x035, 0x3B7, 0x3D7, 0x03D,
+ 0x034, 0x3B2, 0x3DC, 0x03E, 0x033, 0x3AD, 0x3E0, 0x040,
+ 0x032, 0x3A9, 0x3E5, 0x041, 0x031, 0x3A4, 0x3E9, 0x042,
+ 0x030, 0x39F, 0x3ED, 0x043, 0x02F, 0x39B, 0x3F2, 0x045,
+ 0x02E, 0x396, 0x3F6, 0x046, 0x02D, 0x391, 0x3FB, 0x047,
+ 0x02C, 0x38C, 0x3FF, 0x049, 0x02B, 0x388, 0x403, 0x04A,
+ 0x02A, 0x383, 0x408, 0x04C, 0x029, 0x37E, 0x40C, 0x04D,
+ 0x028, 0x379, 0x410, 0x04E, 0x027, 0x374, 0x415, 0x050,
+ 0x026, 0x370, 0x419, 0x051, 0x025, 0x36B, 0x41D, 0x053,
+ 0x024, 0x366, 0x421, 0x054, 0x024, 0x361, 0x425, 0x056,
+ 0x023, 0x35C, 0x42A, 0x057, 0x022, 0x357, 0x42E, 0x059,
+ 0x021, 0x353, 0x432, 0x05A, 0x020, 0x34E, 0x436, 0x05C,
+ 0x020, 0x349, 0x43A, 0x05E, 0x01F, 0x344, 0x43E, 0x05F,
+ 0x01E, 0x33F, 0x442, 0x061, 0x01D, 0x33A, 0x446, 0x063,
+ 0x01D, 0x335, 0x44A, 0x064, 0x01C, 0x330, 0x44E, 0x066,
+ 0x01B, 0x32B, 0x452, 0x068, 0x01B, 0x326, 0x455, 0x06A,
+ 0x01A, 0x322, 0x459, 0x06B, 0x019, 0x31D, 0x45D, 0x06D,
+ 0x018, 0x318, 0x461, 0x06F, 0x018, 0x313, 0x465, 0x071,
+ 0x017, 0x30E, 0x468, 0x073, 0x017, 0x309, 0x46C, 0x075,
+ 0x016, 0x304, 0x470, 0x076, 0x015, 0x2FF, 0x473, 0x078,
+ 0x015, 0x2FA, 0x477, 0x07A, 0x014, 0x2F5, 0x47A, 0x07C,
+ 0x014, 0x2F0, 0x47E, 0x07E, 0x013, 0x2EB, 0x481, 0x080,
+ 0x013, 0x2E6, 0x485, 0x082, 0x012, 0x2E1, 0x488, 0x084,
+ 0x011, 0x2DC, 0x48C, 0x086, 0x011, 0x2D8, 0x48F, 0x089,
+ 0x010, 0x2D3, 0x492, 0x08B, 0x010, 0x2CE, 0x496, 0x08D,
+ 0x00F, 0x2C9, 0x499, 0x08F, 0x00F, 0x2C4, 0x49C, 0x091,
+ 0x00F, 0x2BF, 0x49F, 0x093, 0x00E, 0x2BA, 0x4A2, 0x096,
+ 0x00E, 0x2B5, 0x4A6, 0x098, 0x00D, 0x2B0, 0x4A9, 0x09A,
+ 0x00D, 0x2AB, 0x4AC, 0x09C, 0x00C, 0x2A6, 0x4AF, 0x09F,
+ 0x00C, 0x2A2, 0x4B2, 0x0A1, 0x00B, 0x29D, 0x4B5, 0x0A3,
+ 0x00B, 0x298, 0x4B7, 0x0A6, 0x00B, 0x293, 0x4BA, 0x0A8,
+ 0x00A, 0x28E, 0x4BD, 0x0AB, 0x00A, 0x289, 0x4C0, 0x0AD,
+ 0x00A, 0x284, 0x4C3, 0x0AF, 0x009, 0x280, 0x4C5, 0x0B2,
+ 0x009, 0x27B, 0x4C8, 0x0B4, 0x009, 0x276, 0x4CB, 0x0B7,
+ 0x008, 0x271, 0x4CD, 0x0BA, 0x008, 0x26C, 0x4D0, 0x0BC,
+ 0x008, 0x267, 0x4D2, 0x0BF, 0x007, 0x263, 0x4D5, 0x0C1,
+ 0x007, 0x25E, 0x4D7, 0x0C4, 0x007, 0x259, 0x4D9, 0x0C7,
+ 0x006, 0x254, 0x4DC, 0x0C9, 0x006, 0x250, 0x4DE, 0x0CC,
+ 0x006, 0x24B, 0x4E0, 0x0CF, 0x006, 0x246, 0x4E3, 0x0D2,
+ 0x005, 0x241, 0x4E5, 0x0D4, 0x005, 0x23D, 0x4E7, 0x0D7,
+ 0x005, 0x238, 0x4E9, 0x0DA, 0x005, 0x233, 0x4EB, 0x0DD,
+ 0x004, 0x22F, 0x4ED, 0x0E0, 0x004, 0x22A, 0x4EF, 0x0E3,
+ 0x004, 0x226, 0x4F1, 0x0E6, 0x004, 0x221, 0x4F3, 0x0E9,
+ 0x004, 0x21C, 0x4F5, 0x0EC, 0x003, 0x218, 0x4F6, 0x0EF,
+ 0x003, 0x213, 0x4F8, 0x0F2, 0x003, 0x20F, 0x4FA, 0x0F5,
+ 0x003, 0x20A, 0x4FB, 0x0F8, 0x003, 0x205, 0x4FD, 0x0FB,
+ 0x002, 0x201, 0x4FF, 0x0FE, 0x002, 0x1FC, 0x500, 0x101,
+ 0x002, 0x1F8, 0x502, 0x104, 0x002, 0x1F3, 0x503, 0x107,
+ 0x002, 0x1EF, 0x504, 0x10B, 0x002, 0x1EB, 0x506, 0x10E,
+ 0x002, 0x1E6, 0x507, 0x111, 0x001, 0x1E2, 0x508, 0x114,
+ 0x001, 0x1DD, 0x50A, 0x118, 0x001, 0x1D9, 0x50B, 0x11B,
+ 0x001, 0x1D5, 0x50C, 0x11E, 0x001, 0x1D0, 0x50D, 0x122,
+ 0x001, 0x1CC, 0x50E, 0x125, 0x001, 0x1C8, 0x50F, 0x129,
+ 0x001, 0x1C3, 0x510, 0x12C, 0x001, 0x1BF, 0x511, 0x130,
+ 0x001, 0x1BB, 0x511, 0x133, 0x001, 0x1B7, 0x512, 0x137,
+ 0x000, 0x1B2, 0x513, 0x13A, 0x000, 0x1AE, 0x514, 0x13E,
+ 0x000, 0x1AA, 0x514, 0x141, 0x000, 0x1A6, 0x515, 0x145,
+ 0x000, 0x1A2, 0x516, 0x148, 0x000, 0x19E, 0x516, 0x14C,
+ 0x000, 0x19A, 0x517, 0x150, 0x000, 0x195, 0x517, 0x153,
+ 0x000, 0x191, 0x517, 0x157, 0x000, 0x18D, 0x518, 0x15B,
+ 0x000, 0x189, 0x518, 0x15F, 0x000, 0x185, 0x518, 0x162,
+ 0x000, 0x181, 0x518, 0x166, 0x000, 0x17D, 0x518, 0x16A,
+ 0x000, 0x17A, 0x519, 0x16E, 0x000, 0x176, 0x519, 0x172};
+#endif
diff --git a/plugins/dfsound/nullsnd.c b/plugins/dfsound/nullsnd.c new file mode 100644 index 0000000..bf07909 --- /dev/null +++ b/plugins/dfsound/nullsnd.c @@ -0,0 +1,24 @@ +#include "stdafx.h" +#define _IN_OSS +#include "externals.h" + +// SETUP SOUND +void SetupSound(void) +{ +} + +// REMOVE SOUND +void RemoveSound(void) +{ +} + +// GET BYTES BUFFERED +unsigned long SoundGetBytesBuffered(void) +{ + return 0; +} + +// FEED SOUND DATA +void SoundFeedStreamData(unsigned char* pSound,long lBytes) +{ +} diff --git a/plugins/dfsound/oss.c b/plugins/dfsound/oss.c new file mode 100644 index 0000000..f4dd215 --- /dev/null +++ b/plugins/dfsound/oss.c @@ -0,0 +1,159 @@ +/*************************************************************************** + oss.c - description + ------------------- + begin : Wed May 15 2002 + copyright : (C) 2002 by Pete Bernert + email : BlackDove@addcom.de + ***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +#include "stdafx.h" + +#define _IN_OSS + +#include "externals.h" + +//////////////////////////////////////////////////////////////////////// +// oss globals +//////////////////////////////////////////////////////////////////////// + +#define OSS_MODE_STEREO 1 +#define OSS_MODE_MONO 0 + +#define OSS_SPEED_44100 44100 + +static int oss_audio_fd = -1; +extern int errno; + +//////////////////////////////////////////////////////////////////////// +// SETUP SOUND +//////////////////////////////////////////////////////////////////////// + +void SetupSound(void) +{ + int pspeed=44100; + int pstereo; + int format; + int fragsize = 0; + int myfrag; + int oss_speed, oss_stereo; + + if(iDisStereo) pstereo=OSS_MODE_MONO; + else pstereo=OSS_MODE_STEREO; + + oss_speed = pspeed; + oss_stereo = pstereo; + + if((oss_audio_fd=open("/dev/dsp",O_WRONLY,0))==-1) + { + printf("Sound device not available!\n"); + return; + } + + if(ioctl(oss_audio_fd,SNDCTL_DSP_RESET,0)==-1) + { + printf("Sound reset failed\n"); + return; + } + + // we use 64 fragments with 1024 bytes each + + fragsize=10; + myfrag=(63<<16)|fragsize; + + if(ioctl(oss_audio_fd,SNDCTL_DSP_SETFRAGMENT,&myfrag)==-1) + { + printf("Sound set fragment failed!\n"); + return; + } + + format = AFMT_S16_NE; + + if(ioctl(oss_audio_fd,SNDCTL_DSP_SETFMT,&format) == -1) + { + printf("Sound format not supported!\n"); + return; + } + + if(format!=AFMT_S16_NE) + { + printf("Sound format not supported!\n"); + return; + } + + if(ioctl(oss_audio_fd,SNDCTL_DSP_STEREO,&oss_stereo)==-1) + { + printf("Stereo mode not supported!\n"); + return; + } + + if(oss_stereo!=1) + { + iDisStereo=1; + } + + if(ioctl(oss_audio_fd,SNDCTL_DSP_SPEED,&oss_speed)==-1) + { + printf("Sound frequency not supported\n"); + return; + } + + if(oss_speed!=pspeed) + { + printf("Sound frequency not supported\n"); + return; + } +} + +//////////////////////////////////////////////////////////////////////// +// REMOVE SOUND +//////////////////////////////////////////////////////////////////////// + +void RemoveSound(void) +{ + if(oss_audio_fd != -1 ) + { + close(oss_audio_fd); + oss_audio_fd = -1; + } +} + +//////////////////////////////////////////////////////////////////////// +// GET BYTES BUFFERED +//////////////////////////////////////////////////////////////////////// + +unsigned long SoundGetBytesBuffered(void) +{ + audio_buf_info info; + unsigned long l; + + if(oss_audio_fd == -1) return SOUNDSIZE; + if(ioctl(oss_audio_fd,SNDCTL_DSP_GETOSPACE,&info)==-1) + l=0; + else + { + if(info.fragments<(info.fragstotal>>1)) // can we write in at least the half of fragments? + l=SOUNDSIZE; // -> no? wait + else l=0; // -> else go on + } + + return l; +} + +//////////////////////////////////////////////////////////////////////// +// FEED SOUND DATA +//////////////////////////////////////////////////////////////////////// + +void SoundFeedStreamData(unsigned char* pSound,long lBytes) +{ + if(oss_audio_fd == -1) return; + write(oss_audio_fd,pSound,lBytes); +} diff --git a/plugins/dfsound/psemuxa.h b/plugins/dfsound/psemuxa.h new file mode 100644 index 0000000..84c6260 --- /dev/null +++ b/plugins/dfsound/psemuxa.h @@ -0,0 +1,28 @@ +//============================================
+//=== Audio XA decoding
+//=== Kazzuya
+//============================================
+
+#ifndef DECODEXA_H
+#define DECODEXA_H
+
+typedef struct
+{
+ long y0, y1;
+} ADPCM_Decode_t;
+
+typedef struct
+{
+ int freq;
+ int nbits;
+ int stereo;
+ int nsamples;
+ ADPCM_Decode_t left, right;
+ short pcm[16384];
+} xa_decode_t;
+
+long xa_decode_sector( xa_decode_t *xdp,
+ unsigned char *sectorp,
+ int is_first_sector );
+
+#endif
diff --git a/plugins/dfsound/pulseaudio.c b/plugins/dfsound/pulseaudio.c new file mode 100644 index 0000000..6005155 --- /dev/null +++ b/plugins/dfsound/pulseaudio.c @@ -0,0 +1,354 @@ +/*************************************************************************** + pulseaudio.c - description + ------------------- +begin : Thu Feb 04 2010 +copyright : (C) 2010 by Tristin Celestin +email : cetris1@umbc.edu +comment : Much of this was taken from simple.c, in the pulseaudio + library +***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +#include "stdafx.h" + +#ifdef USEPULSEAUDIO + +#define _IN_OSS + +#include "externals.h" +#include <pulse/pulseaudio.h> + +//////////////////////////////////////////////////////////////////////// +// pulseaudio structs +//////////////////////////////////////////////////////////////////////// + +typedef struct { + pa_threaded_mainloop *mainloop; + pa_context *context; + pa_mainloop_api *api; + pa_stream *stream; + pa_sample_spec spec; + int first; +} Device; + +typedef struct { + unsigned int frequency; + unsigned int latency_in_msec; +} Settings; + +//////////////////////////////////////////////////////////////////////// +// pulseaudio globals +//////////////////////////////////////////////////////////////////////// + +static Device device = { + .mainloop = NULL, + .api = NULL, + .context = NULL, + .stream = NULL +}; + +static Settings settings = { + .frequency = 44100, + .latency_in_msec = 20, +}; + +// the number of bytes written in SoundFeedStreamData +const int mixlen = 3240; + +// used to calculate how much space is used in the buffer, for debugging purposes +//int maxlength = 0; + +//////////////////////////////////////////////////////////////////////// +// CALLBACKS FOR THREADED MAINLOOP +//////////////////////////////////////////////////////////////////////// +static void context_state_cb (pa_context *context, void *userdata) +{ + Device *dev = userdata; + + if ((context == NULL) || (dev == NULL)) + return; + + switch (pa_context_get_state (context)) + { + case PA_CONTEXT_READY: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + pa_threaded_mainloop_signal (dev->mainloop, 0); + break; + + case PA_CONTEXT_UNCONNECTED: + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + break; + } +} + +static void stream_state_cb (pa_stream *stream, void * userdata) +{ + Device *dev = userdata; + + if ((stream == NULL) || (dev == NULL)) + return; + + switch (pa_stream_get_state (stream)) + { + case PA_STREAM_READY: + case PA_STREAM_FAILED: + case PA_STREAM_TERMINATED: + pa_threaded_mainloop_signal (dev->mainloop, 0); + break; + + case PA_STREAM_UNCONNECTED: + case PA_STREAM_CREATING: + break; + } +} + +static void stream_latency_update_cb (pa_stream *stream, void *userdata) +{ + Device *dev = userdata; + + if ((stream == NULL) || (dev == NULL)) + return; + + pa_threaded_mainloop_signal (dev->mainloop, 0); +} + +static void stream_request_cb (pa_stream *stream, size_t length, void *userdata) +{ + Device *dev = userdata; + + if ((stream == NULL) || (dev == NULL)) + return; + pa_threaded_mainloop_signal (dev->mainloop, 0); +} + +//////////////////////////////////////////////////////////////////////// +// SETUP SOUND +//////////////////////////////////////////////////////////////////////// + +void SetupSound (void) +{ + int error_number; + + // Acquire mainloop /////////////////////////////////////////////////////// + device.mainloop = pa_threaded_mainloop_new (); + if (device.mainloop == NULL) + { + fprintf (stderr, "Could not acquire PulseAudio main loop\n"); + return; + } + + // Acquire context //////////////////////////////////////////////////////// + device.api = pa_threaded_mainloop_get_api (device.mainloop); + device.context = pa_context_new (device.api, "PCSX"); + pa_context_set_state_callback (device.context, context_state_cb, &device); + + if (device.context == NULL) + { + fprintf (stderr, "Could not acquire PulseAudio device context\n"); + return; + } + + // Connect to PulseAudio server /////////////////////////////////////////// + if (pa_context_connect (device.context, NULL, 0, NULL) < 0) + { + error_number = pa_context_errno (device.context); + fprintf (stderr, "Could not connect to PulseAudio server: %s\n", pa_strerror(error_number)); + return; + } + + // Run mainloop until sever context is ready ////////////////////////////// + pa_threaded_mainloop_lock (device.mainloop); + if (pa_threaded_mainloop_start (device.mainloop) < 0) + { + fprintf (stderr, "Could not start mainloop\n"); + return; + } + + pa_context_state_t context_state; + context_state = pa_context_get_state (device.context); + while (context_state != PA_CONTEXT_READY) + { + context_state = pa_context_get_state (device.context); + if (! PA_CONTEXT_IS_GOOD (context_state)) + { + error_number = pa_context_errno (device.context); + fprintf (stderr, "Context state is not good: %s\n", pa_strerror (error_number)); + return; + } + else if (context_state == PA_CONTEXT_READY) + break; + else + fprintf (stderr, "PulseAudio context state is %d\n", context_state); + pa_threaded_mainloop_wait (device.mainloop); + } + + // Set sample spec //////////////////////////////////////////////////////// + device.spec.format = PA_SAMPLE_S16NE; + if (iDisStereo) + device.spec.channels = 1; + else + device.spec.channels = 2; + device.spec.rate = settings.frequency; + + pa_buffer_attr buffer_attributes; + buffer_attributes.tlength = pa_bytes_per_second (& device.spec) / 5; + buffer_attributes.maxlength = buffer_attributes.tlength * 3; + buffer_attributes.minreq = buffer_attributes.tlength / 3; + buffer_attributes.prebuf = buffer_attributes.tlength; + + //maxlength = buffer_attributes.maxlength; + //fprintf (stderr, "Total space: %u\n", buffer_attributes.maxlength); + //fprintf (stderr, "Minimum request size: %u\n", buffer_attributes.minreq); + //fprintf (stderr, "Bytes needed before playback: %u\n", buffer_attributes.prebuf); + //fprintf (stderr, "Target buffer size: %lu\n", buffer_attributes.tlength); + + // Acquire new stream using spec ////////////////////////////////////////// + device.stream = pa_stream_new (device.context, "PCSX", &device.spec, NULL); + if (device.stream == NULL) + { + error_number = pa_context_errno (device.context); + fprintf (stderr, "Could not acquire new PulseAudio stream: %s\n", pa_strerror (error_number)); + return; + } + + // Set callbacks for server events //////////////////////////////////////// + pa_stream_set_state_callback (device.stream, stream_state_cb, &device); + pa_stream_set_write_callback (device.stream, stream_request_cb, &device); + pa_stream_set_latency_update_callback (device.stream, stream_latency_update_cb, &device); + + // Ready stream for playback ////////////////////////////////////////////// + pa_stream_flags_t flags = (pa_stream_flags_t) (PA_STREAM_ADJUST_LATENCY | PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE); + //pa_stream_flags_t flags = (pa_stream_flags_t) (PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_EARLY_REQUESTS); + if (pa_stream_connect_playback (device.stream, NULL, &buffer_attributes, flags, NULL, NULL) < 0) + { + pa_context_errno (device.context); + fprintf (stderr, "Could not connect for playback: %s\n", pa_strerror (error_number)); + return; + } + + // Run mainloop until stream is ready ///////////////////////////////////// + pa_stream_state_t stream_state; + stream_state = pa_stream_get_state (device.stream); + while (stream_state != PA_STREAM_READY) + { + stream_state = pa_stream_get_state (device.stream); + + if (stream_state == PA_STREAM_READY) + break; + + else if (! PA_STREAM_IS_GOOD (stream_state)) + { + error_number = pa_context_errno (device.context); + fprintf (stderr, "Stream state is not good: %s\n", pa_strerror (error_number)); + return; + } + else + fprintf (stderr, "PulseAudio stream state is %d\n", stream_state); + pa_threaded_mainloop_wait (device.mainloop); + } + + pa_threaded_mainloop_unlock (device.mainloop); + + fprintf (stderr, "PulseAudio should be connected\n"); + return; +} + +//////////////////////////////////////////////////////////////////////// +// REMOVE SOUND +//////////////////////////////////////////////////////////////////////// +void RemoveSound (void) +{ + if (device.mainloop != NULL) + pa_threaded_mainloop_stop (device.mainloop); + + // Release in reverse order of acquisition + if (device.stream != NULL) + { + pa_stream_unref (device.stream); + device.stream = NULL; + + } + if (device.context != NULL) + { + pa_context_disconnect (device.context); + pa_context_unref (device.context); + device.context = NULL; + } + + if (device.mainloop != NULL) + { + pa_threaded_mainloop_free (device.mainloop); + device.mainloop = NULL; + } + +} + +//////////////////////////////////////////////////////////////////////// +// GET BYTES BUFFERED +//////////////////////////////////////////////////////////////////////// + +unsigned long SoundGetBytesBuffered (void) +{ + int free_space; + int error_code; + long latency; + int playing = 0; + + if ((device.mainloop == NULL) || (device.api == NULL) || ( device.context == NULL) || (device.stream == NULL)) + return SOUNDSIZE; + + pa_threaded_mainloop_lock (device.mainloop); + free_space = pa_stream_writable_size (device.stream); + pa_threaded_mainloop_unlock (device.mainloop); + + //fprintf (stderr, "Free space: %d\n", free_space); + //fprintf (stderr, "Used space: %d\n", maxlength - free_space); + if (free_space < mixlen * 3) + { + // Don't buffer anymore, just play + //fprintf (stderr, "Not buffering.\n"); + return SOUNDSIZE; + } + else + { + // Buffer some sound + //fprintf (stderr, "Buffering.\n"); + return 0; + } +} + +//////////////////////////////////////////////////////////////////////// +// FEED SOUND DATA +//////////////////////////////////////////////////////////////////////// + +void SoundFeedStreamData (unsigned char *pSound, long lBytes) +{ + int error_code; + int size; + + if (device.mainloop != NULL) + { + pa_threaded_mainloop_lock (device.mainloop); + if (pa_stream_write (device.stream, pSound, lBytes, NULL, 0LL, PA_SEEK_RELATIVE) < 0) + { + fprintf (stderr, "Could not perform write\n"); + } + else + { + //fprintf (stderr, "Wrote %d bytes\n", lBytes); + pa_threaded_mainloop_unlock (device.mainloop); + } + } +} +#endif diff --git a/plugins/dfsound/registers.c b/plugins/dfsound/registers.c new file mode 100644 index 0000000..b684914 --- /dev/null +++ b/plugins/dfsound/registers.c @@ -0,0 +1,589 @@ +/***************************************************************************
+ registers.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_REGISTERS
+
+#include "externals.h"
+#include "registers.h"
+#include "regs.h"
+#include "reverb.h"
+
+/*
+// adsr time values (in ms) by James Higgs ... see the end of
+// the adsr.c source for details
+
+#define ATTACK_MS 514L
+#define DECAYHALF_MS 292L
+#define DECAY_MS 584L
+#define SUSTAIN_MS 450L
+#define RELEASE_MS 446L
+*/
+
+// we have a timebase of 1.020408f ms, not 1 ms... so adjust adsr defines
+#define ATTACK_MS 494L
+#define DECAYHALF_MS 286L
+#define DECAY_MS 572L
+#define SUSTAIN_MS 441L
+#define RELEASE_MS 437L
+
+////////////////////////////////////////////////////////////////////////
+// WRITE REGISTERS: called by main emu
+////////////////////////////////////////////////////////////////////////
+
+void CALLBACK SPUwriteRegister(unsigned long reg, unsigned short val)
+{
+ const unsigned long r=reg&0xfff;
+ regArea[(r-0xc00)>>1] = val;
+
+ if(r>=0x0c00 && r<0x0d80) // some channel info?
+ {
+ int ch=(r>>4)-0xc0; // calc channel
+ switch(r&0x0f)
+ {
+ //------------------------------------------------// r volume
+ case 0:
+ SetVolumeL((unsigned char)ch,val);
+ break;
+ //------------------------------------------------// l volume
+ case 2:
+ SetVolumeR((unsigned char)ch,val);
+ break;
+ //------------------------------------------------// pitch
+ case 4:
+ SetPitch(ch,val);
+ break;
+ //------------------------------------------------// start
+ case 6:
+ s_chan[ch].pStart=spuMemC+((unsigned long) val<<3);
+ break;
+ //------------------------------------------------// level with pre-calcs
+ case 8:
+ {
+ const unsigned long lval=val;unsigned long lx;
+ //---------------------------------------------//
+ s_chan[ch].ADSRX.AttackModeExp=(lval&0x8000)?1:0;
+ s_chan[ch].ADSRX.AttackRate=(lval>>8) & 0x007f;
+ s_chan[ch].ADSRX.DecayRate=(lval>>4) & 0x000f;
+ s_chan[ch].ADSRX.SustainLevel=lval & 0x000f;
+ //---------------------------------------------//
+ if(!iDebugMode) break;
+ //---------------------------------------------// stuff below is only for debug mode
+
+ s_chan[ch].ADSR.AttackModeExp=(lval&0x8000)?1:0; //0x007f
+
+ lx=(((lval>>8) & 0x007f)>>2); // attack time to run from 0 to 100% volume
+ lx=min(31,lx); // no overflow on shift!
+ if(lx)
+ {
+ lx = (1<<lx);
+ if(lx<2147483) lx=(lx*ATTACK_MS)/10000L; // another overflow check
+ else lx=(lx/10000L)*ATTACK_MS;
+ if(!lx) lx=1;
+ }
+ s_chan[ch].ADSR.AttackTime=lx;
+
+ s_chan[ch].ADSR.SustainLevel= // our adsr vol runs from 0 to 1024, so scale the sustain level
+ (1024*((lval) & 0x000f))/15;
+
+ lx=(lval>>4) & 0x000f; // decay:
+ if(lx) // our const decay value is time it takes from 100% to 0% of volume
+ {
+ lx = ((1<<(lx))*DECAY_MS)/10000L;
+ if(!lx) lx=1;
+ }
+ s_chan[ch].ADSR.DecayTime = // so calc how long does it take to run from 100% to the wanted sus level
+ (lx*(1024-s_chan[ch].ADSR.SustainLevel))/1024;
+ }
+ break;
+ //------------------------------------------------// adsr times with pre-calcs
+ case 10:
+ {
+ const unsigned long lval=val;unsigned long lx;
+
+ //----------------------------------------------//
+ s_chan[ch].ADSRX.SustainModeExp = (lval&0x8000)?1:0;
+ s_chan[ch].ADSRX.SustainIncrease= (lval&0x4000)?0:1;
+ s_chan[ch].ADSRX.SustainRate = (lval>>6) & 0x007f;
+ s_chan[ch].ADSRX.ReleaseModeExp = (lval&0x0020)?1:0;
+ s_chan[ch].ADSRX.ReleaseRate = lval & 0x001f;
+ //----------------------------------------------//
+ if(!iDebugMode) break;
+ //----------------------------------------------// stuff below is only for debug mode
+
+ s_chan[ch].ADSR.SustainModeExp = (lval&0x8000)?1:0;
+ s_chan[ch].ADSR.ReleaseModeExp = (lval&0x0020)?1:0;
+
+ lx=((((lval>>6) & 0x007f)>>2)); // sustain time... often very high
+ lx=min(31,lx); // values are used to hold the volume
+ if(lx) // until a sound stop occurs
+ { // the highest value we reach (due to
+ lx = (1<<lx); // overflow checking) is:
+ if(lx<2147483) lx=(lx*SUSTAIN_MS)/10000L; // 94704 seconds = 1578 minutes = 26 hours...
+ else lx=(lx/10000L)*SUSTAIN_MS; // should be enuff... if the stop doesn't
+ if(!lx) lx=1; // come in this time span, I don't care :)
+ }
+ s_chan[ch].ADSR.SustainTime = lx;
+
+ lx=(lval & 0x001f);
+ s_chan[ch].ADSR.ReleaseVal =lx;
+ if(lx) // release time from 100% to 0%
+ { // note: the release time will be
+ lx = (1<<lx); // adjusted when a stop is coming,
+ if(lx<2147483) lx=(lx*RELEASE_MS)/10000L; // so at this time the adsr vol will
+ else lx=(lx/10000L)*RELEASE_MS; // run from (current volume) to 0%
+ if(!lx) lx=1;
+ }
+ s_chan[ch].ADSR.ReleaseTime=lx;
+
+ if(lval & 0x4000) // add/dec flag
+ s_chan[ch].ADSR.SustainModeDec=-1;
+ else s_chan[ch].ADSR.SustainModeDec=1;
+ }
+ break;
+ //------------------------------------------------// adsr volume... mmm have to investigate this
+ case 12:
+ break;
+ //------------------------------------------------//
+ case 14: // loop?
+ //WaitForSingleObject(s_chan[ch].hMutex,2000); // -> no multithread fuckups
+ s_chan[ch].pLoop=spuMemC+((unsigned long) val<<3);
+ s_chan[ch].bIgnoreLoop=1;
+ //ReleaseMutex(s_chan[ch].hMutex); // -> oki, on with the thread
+ break;
+ //------------------------------------------------//
+ }
+ iSpuAsyncWait=0;
+ return;
+ }
+
+ switch(r)
+ {
+ //-------------------------------------------------//
+ case H_SPUaddr:
+ spuAddr = (unsigned long) val<<3;
+ break;
+ //-------------------------------------------------//
+ case H_SPUdata:
+ spuMem[spuAddr>>1] = val;
+ spuAddr+=2;
+ if(spuAddr>0x7ffff) spuAddr=0;
+ break;
+ //-------------------------------------------------//
+ case H_SPUctrl:
+ spuCtrl=val;
+ break;
+ //-------------------------------------------------//
+ case H_SPUstat:
+ spuStat=val & 0xf800;
+ break;
+ //-------------------------------------------------//
+ case H_SPUReverbAddr:
+ if(val==0xFFFF || val<=0x200)
+ {rvb.StartAddr=rvb.CurrAddr=0;}
+ else
+ {
+ const long iv=(unsigned long)val<<2;
+ if(rvb.StartAddr!=iv)
+ {
+ rvb.StartAddr=(unsigned long)val<<2;
+ rvb.CurrAddr=rvb.StartAddr;
+ }
+ }
+ break;
+ //-------------------------------------------------//
+ case H_SPUirqAddr:
+ spuIrq = val;
+ pSpuIrq=spuMemC+((unsigned long) val<<3);
+ break;
+ //-------------------------------------------------//
+ case H_SPUrvolL:
+ rvb.VolLeft=val;
+ break;
+ //-------------------------------------------------//
+ case H_SPUrvolR:
+ rvb.VolRight=val;
+ break;
+ //-------------------------------------------------//
+
+/*
+ case H_ExtLeft:
+ //auxprintf("EL %d\n",val);
+ break;
+ //-------------------------------------------------//
+ case H_ExtRight:
+ //auxprintf("ER %d\n",val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUmvolL:
+ //auxprintf("ML %d\n",val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUmvolR:
+ //auxprintf("MR %d\n",val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUMute1:
+ //auxprintf("M0 %04x\n",val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUMute2:
+ //auxprintf("M1 %04x\n",val);
+ break;
+*/
+ //-------------------------------------------------//
+ case H_SPUon1:
+ SoundOn(0,16,val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUon2:
+ SoundOn(16,24,val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUoff1:
+ SoundOff(0,16,val);
+ break;
+ //-------------------------------------------------//
+ case H_SPUoff2:
+ SoundOff(16,24,val);
+ break;
+ //-------------------------------------------------//
+ case H_CDLeft:
+ iLeftXAVol=val & 0x7fff;
+ if(cddavCallback) cddavCallback(0,val);
+ break;
+ case H_CDRight:
+ iRightXAVol=val & 0x7fff;
+ if(cddavCallback) cddavCallback(1,val);
+ break;
+ //-------------------------------------------------//
+ case H_FMod1:
+ FModOn(0,16,val);
+ break;
+ //-------------------------------------------------//
+ case H_FMod2:
+ FModOn(16,24,val);
+ break;
+ //-------------------------------------------------//
+ case H_Noise1:
+ NoiseOn(0,16,val);
+ break;
+ //-------------------------------------------------//
+ case H_Noise2:
+ NoiseOn(16,24,val);
+ break;
+ //-------------------------------------------------//
+ case H_RVBon1:
+ ReverbOn(0,16,val);
+ break;
+ //-------------------------------------------------//
+ case H_RVBon2:
+ ReverbOn(16,24,val);
+ break;
+ //-------------------------------------------------//
+ case H_Reverb+0:
+
+ rvb.FB_SRC_A=val;
+
+ // OK, here's the fake REVERB stuff...
+ // depending on effect we do more or less delay and repeats... bah
+ // still... better than nothing :)
+
+ SetREVERB(val);
+ break;
+
+
+ case H_Reverb+2 : rvb.FB_SRC_B=(short)val; break;
+ case H_Reverb+4 : rvb.IIR_ALPHA=(short)val; break;
+ case H_Reverb+6 : rvb.ACC_COEF_A=(short)val; break;
+ case H_Reverb+8 : rvb.ACC_COEF_B=(short)val; break;
+ case H_Reverb+10 : rvb.ACC_COEF_C=(short)val; break;
+ case H_Reverb+12 : rvb.ACC_COEF_D=(short)val; break;
+ case H_Reverb+14 : rvb.IIR_COEF=(short)val; break;
+ case H_Reverb+16 : rvb.FB_ALPHA=(short)val; break;
+ case H_Reverb+18 : rvb.FB_X=(short)val; break;
+ case H_Reverb+20 : rvb.IIR_DEST_A0=(short)val; break;
+ case H_Reverb+22 : rvb.IIR_DEST_A1=(short)val; break;
+ case H_Reverb+24 : rvb.ACC_SRC_A0=(short)val; break;
+ case H_Reverb+26 : rvb.ACC_SRC_A1=(short)val; break;
+ case H_Reverb+28 : rvb.ACC_SRC_B0=(short)val; break;
+ case H_Reverb+30 : rvb.ACC_SRC_B1=(short)val; break;
+ case H_Reverb+32 : rvb.IIR_SRC_A0=(short)val; break;
+ case H_Reverb+34 : rvb.IIR_SRC_A1=(short)val; break;
+ case H_Reverb+36 : rvb.IIR_DEST_B0=(short)val; break;
+ case H_Reverb+38 : rvb.IIR_DEST_B1=(short)val; break;
+ case H_Reverb+40 : rvb.ACC_SRC_C0=(short)val; break;
+ case H_Reverb+42 : rvb.ACC_SRC_C1=(short)val; break;
+ case H_Reverb+44 : rvb.ACC_SRC_D0=(short)val; break;
+ case H_Reverb+46 : rvb.ACC_SRC_D1=(short)val; break;
+ case H_Reverb+48 : rvb.IIR_SRC_B1=(short)val; break;
+ case H_Reverb+50 : rvb.IIR_SRC_B0=(short)val; break;
+ case H_Reverb+52 : rvb.MIX_DEST_A0=(short)val; break;
+ case H_Reverb+54 : rvb.MIX_DEST_A1=(short)val; break;
+ case H_Reverb+56 : rvb.MIX_DEST_B0=(short)val; break;
+ case H_Reverb+58 : rvb.MIX_DEST_B1=(short)val; break;
+ case H_Reverb+60 : rvb.IN_COEF_L=(short)val; break;
+ case H_Reverb+62 : rvb.IN_COEF_R=(short)val; break;
+ }
+
+ iSpuAsyncWait=0;
+}
+
+////////////////////////////////////////////////////////////////////////
+// READ REGISTER: called by main emu
+////////////////////////////////////////////////////////////////////////
+
+unsigned short CALLBACK SPUreadRegister(unsigned long reg)
+{
+ const unsigned long r=reg&0xfff;
+
+ iSpuAsyncWait=0;
+
+ if(r>=0x0c00 && r<0x0d80)
+ {
+ switch(r&0x0f)
+ {
+ case 12: // get adsr vol
+ {
+ const int ch=(r>>4)-0xc0;
+ if(s_chan[ch].bNew) return 1; // we are started, but not processed? return 1
+ if(s_chan[ch].ADSRX.lVolume && // same here... we haven't decoded one sample yet, so no envelope yet. return 1 as well
+ !s_chan[ch].ADSRX.EnvelopeVol)
+ return 1;
+ return (unsigned short)(s_chan[ch].ADSRX.EnvelopeVol>>16);
+ }
+
+ case 14: // get loop address
+ {
+ const int ch=(r>>4)-0xc0;
+ if(s_chan[ch].pLoop==NULL) return 0;
+ return (unsigned short)((s_chan[ch].pLoop-spuMemC)>>3);
+ }
+ }
+ }
+
+ switch(r)
+ {
+ case H_SPUctrl:
+ return spuCtrl;
+
+ case H_SPUstat:
+ return spuStat;
+
+ case H_SPUaddr:
+ return (unsigned short)(spuAddr>>3);
+
+ case H_SPUdata:
+ {
+ unsigned short s=spuMem[spuAddr>>1];
+ spuAddr+=2;
+ if(spuAddr>0x7ffff) spuAddr=0;
+ return s;
+ }
+
+ case H_SPUirqAddr:
+ return spuIrq;
+
+ //case H_SPUIsOn1:
+ // return IsSoundOn(0,16);
+
+ //case H_SPUIsOn2:
+ // return IsSoundOn(16,24);
+
+ }
+
+ return regArea[(r-0xc00)>>1];
+}
+
+////////////////////////////////////////////////////////////////////////
+// SOUND ON register write
+////////////////////////////////////////////////////////////////////////
+
+void SoundOn(int start,int end,unsigned short val) // SOUND ON PSX COMAND
+{
+ int ch;
+
+ for(ch=start;ch<end;ch++,val>>=1) // loop channels
+ {
+ if((val&1) && s_chan[ch].pStart) // mmm... start has to be set before key on !?!
+ {
+ s_chan[ch].bIgnoreLoop=0;
+ s_chan[ch].bNew=1;
+ dwNewChannel|=(1<<ch); // bitfield for faster testing
+ }
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// SOUND OFF register write
+////////////////////////////////////////////////////////////////////////
+
+void SoundOff(int start,int end,unsigned short val) // SOUND OFF PSX COMMAND
+{
+ int ch;
+ for(ch=start;ch<end;ch++,val>>=1) // loop channels
+ {
+ if(val&1) // && s_chan[i].bOn) mmm...
+ {
+ s_chan[ch].bStop=1;
+ }
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// FMOD register write
+////////////////////////////////////////////////////////////////////////
+
+void FModOn(int start,int end,unsigned short val) // FMOD ON PSX COMMAND
+{
+ int ch;
+
+ for(ch=start;ch<end;ch++,val>>=1) // loop channels
+ {
+ if(val&1) // -> fmod on/off
+ {
+ if(ch>0)
+ {
+ s_chan[ch].bFMod=1; // --> sound channel
+ s_chan[ch-1].bFMod=2; // --> freq channel
+ }
+ }
+ else
+ {
+ s_chan[ch].bFMod=0; // --> turn off fmod
+ }
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// NOISE register write
+////////////////////////////////////////////////////////////////////////
+
+void NoiseOn(int start,int end,unsigned short val) // NOISE ON PSX COMMAND
+{
+ int ch;
+
+ for(ch=start;ch<end;ch++,val>>=1) // loop channels
+ {
+ if(val&1) // -> noise on/off
+ {
+ s_chan[ch].bNoise=1;
+ }
+ else
+ {
+ s_chan[ch].bNoise=0;
+ }
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// LEFT VOLUME register write
+////////////////////////////////////////////////////////////////////////
+
+// please note: sweep and phase invert are wrong... but I've never seen
+// them used
+
+void SetVolumeL(unsigned char ch,short vol) // LEFT VOLUME
+{
+ s_chan[ch].iLeftVolRaw=vol;
+
+ if(vol&0x8000) // sweep?
+ {
+ short sInc=1; // -> sweep up?
+ if(vol&0x2000) sInc=-1; // -> or down?
+ if(vol&0x1000) vol^=0xffff; // -> mmm... phase inverted? have to investigate this
+ vol=((vol&0x7f)+1)/2; // -> sweep: 0..127 -> 0..64
+ vol+=vol/(2*sInc); // -> HACK: we don't sweep right now, so we just raise/lower the volume by the half!
+ vol*=128;
+ }
+ else // no sweep:
+ {
+ if(vol&0x4000) // -> mmm... phase inverted? have to investigate this
+ //vol^=0xffff;
+ vol=0x3fff-(vol&0x3fff);
+ }
+
+ vol&=0x3fff;
+ s_chan[ch].iLeftVolume=vol; // store volume
+}
+
+////////////////////////////////////////////////////////////////////////
+// RIGHT VOLUME register write
+////////////////////////////////////////////////////////////////////////
+
+void SetVolumeR(unsigned char ch,short vol) // RIGHT VOLUME
+{
+ s_chan[ch].iRightVolRaw=vol;
+
+ if(vol&0x8000) // comments... see above :)
+ {
+ short sInc=1;
+ if(vol&0x2000) sInc=-1;
+ if(vol&0x1000) vol^=0xffff;
+ vol=((vol&0x7f)+1)/2;
+ vol+=vol/(2*sInc);
+ vol*=128;
+ }
+ else
+ {
+ if(vol&0x4000) //vol=vol^=0xffff;
+ vol=0x3fff-(vol&0x3fff);
+ }
+
+ vol&=0x3fff;
+
+ s_chan[ch].iRightVolume=vol;
+}
+
+////////////////////////////////////////////////////////////////////////
+// PITCH register write
+////////////////////////////////////////////////////////////////////////
+
+void SetPitch(int ch,unsigned short val) // SET PITCH
+{
+ int NP;
+ if(val>0x3fff) NP=0x3fff; // get pitch val
+ else NP=val;
+
+ s_chan[ch].iRawPitch=NP;
+
+ NP=(44100L*NP)/4096L; // calc frequency
+ if(NP<1) NP=1; // some security
+ s_chan[ch].iActFreq=NP; // store frequency
+}
+
+////////////////////////////////////////////////////////////////////////
+// REVERB register write
+////////////////////////////////////////////////////////////////////////
+
+void ReverbOn(int start,int end,unsigned short val) // REVERB ON PSX COMMAND
+{
+ int ch;
+
+ for(ch=start;ch<end;ch++,val>>=1) // loop channels
+ {
+ if(val&1) // -> reverb on/off
+ {
+ s_chan[ch].bReverb=1;
+ }
+ else
+ {
+ s_chan[ch].bReverb=0;
+ }
+ }
+}
diff --git a/plugins/dfsound/registers.h b/plugins/dfsound/registers.h new file mode 100644 index 0000000..f2a9397 --- /dev/null +++ b/plugins/dfsound/registers.h @@ -0,0 +1,144 @@ +/***************************************************************************
+ registers.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#define H_SPUReverbAddr 0x0da2
+#define H_SPUirqAddr 0x0da4
+#define H_SPUaddr 0x0da6
+#define H_SPUdata 0x0da8
+#define H_SPUctrl 0x0daa
+#define H_SPUstat 0x0dae
+#define H_SPUmvolL 0x0d80
+#define H_SPUmvolR 0x0d82
+#define H_SPUrvolL 0x0d84
+#define H_SPUrvolR 0x0d86
+#define H_SPUon1 0x0d88
+#define H_SPUon2 0x0d8a
+#define H_SPUoff1 0x0d8c
+#define H_SPUoff2 0x0d8e
+#define H_FMod1 0x0d90
+#define H_FMod2 0x0d92
+#define H_Noise1 0x0d94
+#define H_Noise2 0x0d96
+#define H_RVBon1 0x0d98
+#define H_RVBon2 0x0d9a
+#define H_SPUMute1 0x0d9c
+#define H_SPUMute2 0x0d9e
+#define H_CDLeft 0x0db0
+#define H_CDRight 0x0db2
+#define H_ExtLeft 0x0db4
+#define H_ExtRight 0x0db6
+#define H_Reverb 0x0dc0
+#define H_SPUPitch0 0x0c04
+#define H_SPUPitch1 0x0c14
+#define H_SPUPitch2 0x0c24
+#define H_SPUPitch3 0x0c34
+#define H_SPUPitch4 0x0c44
+#define H_SPUPitch5 0x0c54
+#define H_SPUPitch6 0x0c64
+#define H_SPUPitch7 0x0c74
+#define H_SPUPitch8 0x0c84
+#define H_SPUPitch9 0x0c94
+#define H_SPUPitch10 0x0ca4
+#define H_SPUPitch11 0x0cb4
+#define H_SPUPitch12 0x0cc4
+#define H_SPUPitch13 0x0cd4
+#define H_SPUPitch14 0x0ce4
+#define H_SPUPitch15 0x0cf4
+#define H_SPUPitch16 0x0d04
+#define H_SPUPitch17 0x0d14
+#define H_SPUPitch18 0x0d24
+#define H_SPUPitch19 0x0d34
+#define H_SPUPitch20 0x0d44
+#define H_SPUPitch21 0x0d54
+#define H_SPUPitch22 0x0d64
+#define H_SPUPitch23 0x0d74
+
+#define H_SPUStartAdr0 0x0c06
+#define H_SPUStartAdr1 0x0c16
+#define H_SPUStartAdr2 0x0c26
+#define H_SPUStartAdr3 0x0c36
+#define H_SPUStartAdr4 0x0c46
+#define H_SPUStartAdr5 0x0c56
+#define H_SPUStartAdr6 0x0c66
+#define H_SPUStartAdr7 0x0c76
+#define H_SPUStartAdr8 0x0c86
+#define H_SPUStartAdr9 0x0c96
+#define H_SPUStartAdr10 0x0ca6
+#define H_SPUStartAdr11 0x0cb6
+#define H_SPUStartAdr12 0x0cc6
+#define H_SPUStartAdr13 0x0cd6
+#define H_SPUStartAdr14 0x0ce6
+#define H_SPUStartAdr15 0x0cf6
+#define H_SPUStartAdr16 0x0d06
+#define H_SPUStartAdr17 0x0d16
+#define H_SPUStartAdr18 0x0d26
+#define H_SPUStartAdr19 0x0d36
+#define H_SPUStartAdr20 0x0d46
+#define H_SPUStartAdr21 0x0d56
+#define H_SPUStartAdr22 0x0d66
+#define H_SPUStartAdr23 0x0d76
+
+#define H_SPULoopAdr0 0x0c0e
+#define H_SPULoopAdr1 0x0c1e
+#define H_SPULoopAdr2 0x0c2e
+#define H_SPULoopAdr3 0x0c3e
+#define H_SPULoopAdr4 0x0c4e
+#define H_SPULoopAdr5 0x0c5e
+#define H_SPULoopAdr6 0x0c6e
+#define H_SPULoopAdr7 0x0c7e
+#define H_SPULoopAdr8 0x0c8e
+#define H_SPULoopAdr9 0x0c9e
+#define H_SPULoopAdr10 0x0cae
+#define H_SPULoopAdr11 0x0cbe
+#define H_SPULoopAdr12 0x0cce
+#define H_SPULoopAdr13 0x0cde
+#define H_SPULoopAdr14 0x0cee
+#define H_SPULoopAdr15 0x0cfe
+#define H_SPULoopAdr16 0x0d0e
+#define H_SPULoopAdr17 0x0d1e
+#define H_SPULoopAdr18 0x0d2e
+#define H_SPULoopAdr19 0x0d3e
+#define H_SPULoopAdr20 0x0d4e
+#define H_SPULoopAdr21 0x0d5e
+#define H_SPULoopAdr22 0x0d6e
+#define H_SPULoopAdr23 0x0d7e
+
+#define H_SPU_ADSRLevel0 0x0c08
+#define H_SPU_ADSRLevel1 0x0c18
+#define H_SPU_ADSRLevel2 0x0c28
+#define H_SPU_ADSRLevel3 0x0c38
+#define H_SPU_ADSRLevel4 0x0c48
+#define H_SPU_ADSRLevel5 0x0c58
+#define H_SPU_ADSRLevel6 0x0c68
+#define H_SPU_ADSRLevel7 0x0c78
+#define H_SPU_ADSRLevel8 0x0c88
+#define H_SPU_ADSRLevel9 0x0c98
+#define H_SPU_ADSRLevel10 0x0ca8
+#define H_SPU_ADSRLevel11 0x0cb8
+#define H_SPU_ADSRLevel12 0x0cc8
+#define H_SPU_ADSRLevel13 0x0cd8
+#define H_SPU_ADSRLevel14 0x0ce8
+#define H_SPU_ADSRLevel15 0x0cf8
+#define H_SPU_ADSRLevel16 0x0d08
+#define H_SPU_ADSRLevel17 0x0d18
+#define H_SPU_ADSRLevel18 0x0d28
+#define H_SPU_ADSRLevel19 0x0d38
+#define H_SPU_ADSRLevel20 0x0d48
+#define H_SPU_ADSRLevel21 0x0d58
+#define H_SPU_ADSRLevel22 0x0d68
+#define H_SPU_ADSRLevel23 0x0d78
+
diff --git a/plugins/dfsound/regs.h b/plugins/dfsound/regs.h new file mode 100644 index 0000000..3d2689b --- /dev/null +++ b/plugins/dfsound/regs.h @@ -0,0 +1,27 @@ +/***************************************************************************
+ regs.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+void SoundOn(int start,int end,unsigned short val);
+void SoundOff(int start,int end,unsigned short val);
+void FModOn(int start,int end,unsigned short val);
+void NoiseOn(int start,int end,unsigned short val);
+void SetVolumeL(unsigned char ch,short vol);
+void SetVolumeR(unsigned char ch,short vol);
+void SetPitch(int ch,unsigned short val);
+void ReverbOn(int start,int end,unsigned short val);
+void CALLBACK SPUwriteRegister(unsigned long reg, unsigned short val);
+
diff --git a/plugins/dfsound/reverb.c b/plugins/dfsound/reverb.c new file mode 100644 index 0000000..92e31fc --- /dev/null +++ b/plugins/dfsound/reverb.c @@ -0,0 +1,462 @@ +/***************************************************************************
+ reverb.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_REVERB
+
+// will be included from spu.c
+#ifdef _IN_SPU
+
+////////////////////////////////////////////////////////////////////////
+// globals
+////////////////////////////////////////////////////////////////////////
+
+// REVERB info and timing vars...
+
+int * sRVBPlay = 0;
+int * sRVBEnd = 0;
+int * sRVBStart = 0;
+int iReverbOff = -1; // some delay factor for reverb
+int iReverbRepeat = 0;
+int iReverbNum = 1;
+
+////////////////////////////////////////////////////////////////////////
+// SET REVERB
+////////////////////////////////////////////////////////////////////////
+
+void SetREVERB(unsigned short val)
+{
+ switch(val)
+ {
+ case 0x0000: iReverbOff=-1; break; // off
+ case 0x007D: iReverbOff=32; iReverbNum=2; iReverbRepeat=128; break; // ok room
+
+ case 0x0033: iReverbOff=32; iReverbNum=2; iReverbRepeat=64; break; // studio small
+ case 0x00B1: iReverbOff=48; iReverbNum=2; iReverbRepeat=96; break; // ok studio medium
+ case 0x00E3: iReverbOff=64; iReverbNum=2; iReverbRepeat=128; break; // ok studio large ok
+
+ case 0x01A5: iReverbOff=128; iReverbNum=4; iReverbRepeat=32; break; // ok hall
+ case 0x033D: iReverbOff=256; iReverbNum=4; iReverbRepeat=64; break; // space echo
+ case 0x0001: iReverbOff=184; iReverbNum=3; iReverbRepeat=128; break; // echo/delay
+ case 0x0017: iReverbOff=128; iReverbNum=2; iReverbRepeat=128; break; // half echo
+ default: iReverbOff=32; iReverbNum=1; iReverbRepeat=0; break;
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// START REVERB
+////////////////////////////////////////////////////////////////////////
+
+INLINE void StartREVERB(int ch)
+{
+ if(s_chan[ch].bReverb && (spuCtrl&0x80)) // reverb possible?
+ {
+ if(iUseReverb==2) s_chan[ch].bRVBActive=1;
+ else
+ if(iUseReverb==1 && iReverbOff>0) // -> fake reverb used?
+ {
+ s_chan[ch].bRVBActive=1; // -> activate it
+ s_chan[ch].iRVBOffset=iReverbOff*45;
+ s_chan[ch].iRVBRepeat=iReverbRepeat*45;
+ s_chan[ch].iRVBNum =iReverbNum;
+ }
+ }
+ else s_chan[ch].bRVBActive=0; // else -> no reverb
+}
+
+////////////////////////////////////////////////////////////////////////
+// HELPER FOR NEILL'S REVERB: re-inits our reverb mixing buf
+////////////////////////////////////////////////////////////////////////
+
+INLINE void InitREVERB(void)
+{
+ if(iUseReverb==2)
+ {memset(sRVBStart,0,NSSIZE*2*4);}
+}
+
+////////////////////////////////////////////////////////////////////////
+// STORE REVERB
+////////////////////////////////////////////////////////////////////////
+
+INLINE void StoreREVERB(int ch,int ns)
+{
+ if(iUseReverb==0) return;
+ else
+ if(iUseReverb==2) // -------------------------------- // Neil's reverb
+ {
+ const int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000;
+ const int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000;
+
+ ns<<=1;
+
+ *(sRVBStart+ns) +=iRxl; // -> we mix all active reverb channels into an extra buffer
+ *(sRVBStart+ns+1)+=iRxr;
+ }
+ else // --------------------------------------------- // Pete's easy fake reverb
+ {
+ int * pN;int iRn,iRr=0;
+
+ // we use the half channel volume (/0x8000) for the first reverb effects, quarter for next and so on
+
+ int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x8000;
+ int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x8000;
+
+ for(iRn=1;iRn<=s_chan[ch].iRVBNum;iRn++,iRr+=s_chan[ch].iRVBRepeat,iRxl/=2,iRxr/=2)
+ {
+ pN=sRVBPlay+((s_chan[ch].iRVBOffset+iRr+ns)<<1);
+ if(pN>=sRVBEnd) pN=sRVBStart+(pN-sRVBEnd);
+
+ (*pN)+=iRxl;
+ pN++;
+ (*pN)+=iRxr;
+ }
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int g_buffer(int iOff) // get_buffer content helper: takes care about wraps
+{
+ short * p=(short *)spuMem;
+ iOff=(iOff*4)+rvb.CurrAddr;
+ while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);
+ while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);
+ return (int)*(p+iOff);
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE void s_buffer(int iOff,int iVal) // set_buffer content helper: takes care about wraps and clipping
+{
+ short * p=(short *)spuMem;
+ iOff=(iOff*4)+rvb.CurrAddr;
+ while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);
+ while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);
+ if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L;
+ *(p+iOff)=(short)iVal;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE void s_buffer1(int iOff,int iVal) // set_buffer (+1 sample) content helper: takes care about wraps and clipping
+{
+ short * p=(short *)spuMem;
+ iOff=(iOff*4)+rvb.CurrAddr+1;
+ while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);
+ while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);
+ if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L;
+ *(p+iOff)=(short)iVal;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int MixREVERBLeft(int ns)
+{
+ if(iUseReverb==0) return 0;
+ else
+ if(iUseReverb==2)
+ {
+ static int iCnt=0; // this func will be called with 44.1 khz
+
+ if(!rvb.StartAddr) // reverb is off
+ {
+ rvb.iLastRVBLeft=rvb.iLastRVBRight=rvb.iRVBLeft=rvb.iRVBRight=0;
+ return 0;
+ }
+
+ iCnt++;
+
+ if(iCnt&1) // we work on every second left value: downsample to 22 khz
+ {
+ if(spuCtrl&0x80) // -> reverb on? oki
+ {
+ int ACC0,ACC1,FB_A0,FB_A1,FB_B0,FB_B1;
+
+ const int INPUT_SAMPLE_L=*(sRVBStart+(ns<<1));
+ const int INPUT_SAMPLE_R=*(sRVBStart+(ns<<1)+1);
+
+ const int IIR_INPUT_A0 = (g_buffer(rvb.IIR_SRC_A0) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb.IN_COEF_L)/32768L;
+ const int IIR_INPUT_A1 = (g_buffer(rvb.IIR_SRC_A1) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb.IN_COEF_R)/32768L;
+ const int IIR_INPUT_B0 = (g_buffer(rvb.IIR_SRC_B0) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb.IN_COEF_L)/32768L;
+ const int IIR_INPUT_B1 = (g_buffer(rvb.IIR_SRC_B1) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb.IN_COEF_R)/32768L;
+
+ const int IIR_A0 = (IIR_INPUT_A0 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_A0) * (32768L - rvb.IIR_ALPHA))/32768L;
+ const int IIR_A1 = (IIR_INPUT_A1 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_A1) * (32768L - rvb.IIR_ALPHA))/32768L;
+ const int IIR_B0 = (IIR_INPUT_B0 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_B0) * (32768L - rvb.IIR_ALPHA))/32768L;
+ const int IIR_B1 = (IIR_INPUT_B1 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_B1) * (32768L - rvb.IIR_ALPHA))/32768L;
+
+ s_buffer1(rvb.IIR_DEST_A0, IIR_A0);
+ s_buffer1(rvb.IIR_DEST_A1, IIR_A1);
+ s_buffer1(rvb.IIR_DEST_B0, IIR_B0);
+ s_buffer1(rvb.IIR_DEST_B1, IIR_B1);
+
+ ACC0 = (g_buffer(rvb.ACC_SRC_A0) * rvb.ACC_COEF_A)/32768L +
+ (g_buffer(rvb.ACC_SRC_B0) * rvb.ACC_COEF_B)/32768L +
+ (g_buffer(rvb.ACC_SRC_C0) * rvb.ACC_COEF_C)/32768L +
+ (g_buffer(rvb.ACC_SRC_D0) * rvb.ACC_COEF_D)/32768L;
+ ACC1 = (g_buffer(rvb.ACC_SRC_A1) * rvb.ACC_COEF_A)/32768L +
+ (g_buffer(rvb.ACC_SRC_B1) * rvb.ACC_COEF_B)/32768L +
+ (g_buffer(rvb.ACC_SRC_C1) * rvb.ACC_COEF_C)/32768L +
+ (g_buffer(rvb.ACC_SRC_D1) * rvb.ACC_COEF_D)/32768L;
+
+ FB_A0 = g_buffer(rvb.MIX_DEST_A0 - rvb.FB_SRC_A);
+ FB_A1 = g_buffer(rvb.MIX_DEST_A1 - rvb.FB_SRC_A);
+ FB_B0 = g_buffer(rvb.MIX_DEST_B0 - rvb.FB_SRC_B);
+ FB_B1 = g_buffer(rvb.MIX_DEST_B1 - rvb.FB_SRC_B);
+
+ s_buffer(rvb.MIX_DEST_A0, ACC0 - (FB_A0 * rvb.FB_ALPHA)/32768L);
+ s_buffer(rvb.MIX_DEST_A1, ACC1 - (FB_A1 * rvb.FB_ALPHA)/32768L);
+
+ s_buffer(rvb.MIX_DEST_B0, (rvb.FB_ALPHA * ACC0)/32768L - (FB_A0 * (int)(rvb.FB_ALPHA^0xFFFF8000))/32768L - (FB_B0 * rvb.FB_X)/32768L);
+ s_buffer(rvb.MIX_DEST_B1, (rvb.FB_ALPHA * ACC1)/32768L - (FB_A1 * (int)(rvb.FB_ALPHA^0xFFFF8000))/32768L - (FB_B1 * rvb.FB_X)/32768L);
+
+ rvb.iLastRVBLeft = rvb.iRVBLeft;
+ rvb.iLastRVBRight = rvb.iRVBRight;
+
+ rvb.iRVBLeft = (g_buffer(rvb.MIX_DEST_A0)+g_buffer(rvb.MIX_DEST_B0))/3;
+ rvb.iRVBRight = (g_buffer(rvb.MIX_DEST_A1)+g_buffer(rvb.MIX_DEST_B1))/3;
+
+ rvb.iRVBLeft = (rvb.iRVBLeft * rvb.VolLeft) / 0x4000;
+ rvb.iRVBRight = (rvb.iRVBRight * rvb.VolRight) / 0x4000;
+
+ rvb.CurrAddr++;
+ if(rvb.CurrAddr>0x3ffff) rvb.CurrAddr=rvb.StartAddr;
+
+ return rvb.iLastRVBLeft+(rvb.iRVBLeft-rvb.iLastRVBLeft)/2;
+ }
+ else // -> reverb off
+ {
+ rvb.iLastRVBLeft=rvb.iLastRVBRight=rvb.iRVBLeft=rvb.iRVBRight=0;
+ }
+
+ rvb.CurrAddr++;
+ if(rvb.CurrAddr>0x3ffff) rvb.CurrAddr=rvb.StartAddr;
+ }
+
+ return rvb.iLastRVBLeft;
+ }
+ else // easy fake reverb:
+ {
+ const int iRV=*sRVBPlay; // -> simply take the reverb mix buf value
+ *sRVBPlay++=0; // -> init it after
+ if(sRVBPlay>=sRVBEnd) sRVBPlay=sRVBStart; // -> and take care about wrap arounds
+ return iRV; // -> return reverb mix buf val
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int MixREVERBRight(void)
+{
+ if(iUseReverb==0) return 0;
+ else
+ if(iUseReverb==2) // Neill's reverb:
+ {
+ int i=rvb.iLastRVBRight+(rvb.iRVBRight-rvb.iLastRVBRight)/2;
+ rvb.iLastRVBRight=rvb.iRVBRight;
+ return i; // -> just return the last right reverb val (little bit scaled by the previous right val)
+ }
+ else // easy fake reverb:
+ {
+ const int iRV=*sRVBPlay; // -> simply take the reverb mix buf value
+ *sRVBPlay++=0; // -> init it after
+ if(sRVBPlay>=sRVBEnd) sRVBPlay=sRVBStart; // -> and take care about wrap arounds
+ return iRV; // -> return reverb mix buf val
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+#endif
+
+/*
+-----------------------------------------------------------------------------
+PSX reverb hardware notes
+by Neill Corlett
+-----------------------------------------------------------------------------
+
+Yadda yadda disclaimer yadda probably not perfect yadda well it's okay anyway
+yadda yadda.
+
+-----------------------------------------------------------------------------
+
+Basics
+------
+
+- The reverb buffer is 22khz 16-bit mono PCM.
+- It starts at the reverb address given by 1DA2, extends to
+ the end of sound RAM, and wraps back to the 1DA2 address.
+
+Setting the address at 1DA2 resets the current reverb work address.
+
+This work address ALWAYS increments every 1/22050 sec., regardless of
+whether reverb is enabled (bit 7 of 1DAA set).
+
+And the contents of the reverb buffer ALWAYS play, scaled by the
+"reverberation depth left/right" volumes (1D84/1D86).
+(which, by the way, appear to be scaled so 3FFF=approx. 1.0, 4000=-1.0)
+
+-----------------------------------------------------------------------------
+
+Register names
+--------------
+
+These are probably not their real names.
+These are probably not even correct names.
+We will use them anyway, because we can.
+
+1DC0: FB_SRC_A (offset)
+1DC2: FB_SRC_B (offset)
+1DC4: IIR_ALPHA (coef.)
+1DC6: ACC_COEF_A (coef.)
+1DC8: ACC_COEF_B (coef.)
+1DCA: ACC_COEF_C (coef.)
+1DCC: ACC_COEF_D (coef.)
+1DCE: IIR_COEF (coef.)
+1DD0: FB_ALPHA (coef.)
+1DD2: FB_X (coef.)
+1DD4: IIR_DEST_A0 (offset)
+1DD6: IIR_DEST_A1 (offset)
+1DD8: ACC_SRC_A0 (offset)
+1DDA: ACC_SRC_A1 (offset)
+1DDC: ACC_SRC_B0 (offset)
+1DDE: ACC_SRC_B1 (offset)
+1DE0: IIR_SRC_A0 (offset)
+1DE2: IIR_SRC_A1 (offset)
+1DE4: IIR_DEST_B0 (offset)
+1DE6: IIR_DEST_B1 (offset)
+1DE8: ACC_SRC_C0 (offset)
+1DEA: ACC_SRC_C1 (offset)
+1DEC: ACC_SRC_D0 (offset)
+1DEE: ACC_SRC_D1 (offset)
+1DF0: IIR_SRC_B1 (offset)
+1DF2: IIR_SRC_B0 (offset)
+1DF4: MIX_DEST_A0 (offset)
+1DF6: MIX_DEST_A1 (offset)
+1DF8: MIX_DEST_B0 (offset)
+1DFA: MIX_DEST_B1 (offset)
+1DFC: IN_COEF_L (coef.)
+1DFE: IN_COEF_R (coef.)
+
+The coefficients are signed fractional values.
+-32768 would be -1.0
+ 32768 would be 1.0 (if it were possible... the highest is of course 32767)
+
+The offsets are (byte/8) offsets into the reverb buffer.
+i.e. you multiply them by 8, you get byte offsets.
+You can also think of them as (samples/4) offsets.
+They appear to be signed. They can be negative.
+None of the documented presets make them negative, though.
+
+Yes, 1DF0 and 1DF2 appear to be backwards. Not a typo.
+
+-----------------------------------------------------------------------------
+
+What it does
+------------
+
+We take all reverb sources:
+- regular channels that have the reverb bit on
+- cd and external sources, if their reverb bits are on
+and mix them into one stereo 44100hz signal.
+
+Lowpass/downsample that to 22050hz. The PSX uses a proper bandlimiting
+algorithm here, but I haven't figured out the hysterically exact specifics.
+I use an 8-tap filter with these coefficients, which are nice but probably
+not the real ones:
+
+0.037828187894
+0.157538631280
+0.321159685278
+0.449322115345
+0.449322115345
+0.321159685278
+0.157538631280
+0.037828187894
+
+So we have two input samples (INPUT_SAMPLE_L, INPUT_SAMPLE_R) every 22050hz.
+
+* IN MY EMULATION, I divide these by 2 to make it clip less.
+ (and of course the L/R output coefficients are adjusted to compensate)
+ The real thing appears to not do this.
+
+At every 22050hz tick:
+- If the reverb bit is enabled (bit 7 of 1DAA), execute the reverb
+ steady-state algorithm described below
+- AFTERWARDS, retrieve the "wet out" L and R samples from the reverb buffer
+ (This part may not be exactly right and I guessed at the coefs. TODO: check later.)
+ L is: 0.333 * (buffer[MIX_DEST_A0] + buffer[MIX_DEST_B0])
+ R is: 0.333 * (buffer[MIX_DEST_A1] + buffer[MIX_DEST_B1])
+- Advance the current buffer position by 1 sample
+
+The wet out L and R are then upsampled to 44100hz and played at the
+"reverberation depth left/right" (1D84/1D86) volume, independent of the main
+volume.
+
+-----------------------------------------------------------------------------
+
+Reverb steady-state
+-------------------
+
+The reverb steady-state algorithm is fairly clever, and of course by
+"clever" I mean "batshit insane".
+
+buffer[x] is relative to the current buffer position, not the beginning of
+the buffer. Note that all buffer offsets must wrap around so they're
+contained within the reverb work area.
+
+Clipping is performed at the end... maybe also sooner, but definitely at
+the end.
+
+IIR_INPUT_A0 = buffer[IIR_SRC_A0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;
+IIR_INPUT_A1 = buffer[IIR_SRC_A1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;
+IIR_INPUT_B0 = buffer[IIR_SRC_B0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;
+IIR_INPUT_B1 = buffer[IIR_SRC_B1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;
+
+IIR_A0 = IIR_INPUT_A0 * IIR_ALPHA + buffer[IIR_DEST_A0] * (1.0 - IIR_ALPHA);
+IIR_A1 = IIR_INPUT_A1 * IIR_ALPHA + buffer[IIR_DEST_A1] * (1.0 - IIR_ALPHA);
+IIR_B0 = IIR_INPUT_B0 * IIR_ALPHA + buffer[IIR_DEST_B0] * (1.0 - IIR_ALPHA);
+IIR_B1 = IIR_INPUT_B1 * IIR_ALPHA + buffer[IIR_DEST_B1] * (1.0 - IIR_ALPHA);
+
+buffer[IIR_DEST_A0 + 1sample] = IIR_A0;
+buffer[IIR_DEST_A1 + 1sample] = IIR_A1;
+buffer[IIR_DEST_B0 + 1sample] = IIR_B0;
+buffer[IIR_DEST_B1 + 1sample] = IIR_B1;
+
+ACC0 = buffer[ACC_SRC_A0] * ACC_COEF_A +
+ buffer[ACC_SRC_B0] * ACC_COEF_B +
+ buffer[ACC_SRC_C0] * ACC_COEF_C +
+ buffer[ACC_SRC_D0] * ACC_COEF_D;
+ACC1 = buffer[ACC_SRC_A1] * ACC_COEF_A +
+ buffer[ACC_SRC_B1] * ACC_COEF_B +
+ buffer[ACC_SRC_C1] * ACC_COEF_C +
+ buffer[ACC_SRC_D1] * ACC_COEF_D;
+
+FB_A0 = buffer[MIX_DEST_A0 - FB_SRC_A];
+FB_A1 = buffer[MIX_DEST_A1 - FB_SRC_A];
+FB_B0 = buffer[MIX_DEST_B0 - FB_SRC_B];
+FB_B1 = buffer[MIX_DEST_B1 - FB_SRC_B];
+
+buffer[MIX_DEST_A0] = ACC0 - FB_A0 * FB_ALPHA;
+buffer[MIX_DEST_A1] = ACC1 - FB_A1 * FB_ALPHA;
+buffer[MIX_DEST_B0] = (FB_ALPHA * ACC0) - FB_A0 * (FB_ALPHA^0x8000) - FB_B0 * FB_X;
+buffer[MIX_DEST_B1] = (FB_ALPHA * ACC1) - FB_A1 * (FB_ALPHA^0x8000) - FB_B1 * FB_X;
+
+-----------------------------------------------------------------------------
+*/
+
diff --git a/plugins/dfsound/reverb.h b/plugins/dfsound/reverb.h new file mode 100644 index 0000000..ce61818 --- /dev/null +++ b/plugins/dfsound/reverb.h @@ -0,0 +1,21 @@ +/***************************************************************************
+ reverb.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+void SetREVERB(unsigned short val);
+INLINE void StartREVERB(int ch);
+INLINE void StoreREVERB(int ch,int ns);
+
diff --git a/plugins/dfsound/sdl.c b/plugins/dfsound/sdl.c new file mode 100644 index 0000000..45ccba2 --- /dev/null +++ b/plugins/dfsound/sdl.c @@ -0,0 +1,135 @@ +/* SDL Driver for P.E.Op.S Sound Plugin + * Copyright (c) 2010, Wei Mingzhi <whistler_wmz@users.sf.net>. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02111-1307 USA + */ + +#include "stdafx.h" + +#include "externals.h" +#include <SDL.h> + +#define BUFFER_SIZE 22050 + +short *pSndBuffer = NULL; +int iBufSize = 0; +volatile int iReadPos = 0, iWritePos = 0; + +static void SOUND_FillAudio(void *unused, Uint8 *stream, int len) { + short *p = (short *)stream; + + len /= sizeof(short); + + while (iReadPos != iWritePos && len > 0) { + *p++ = pSndBuffer[iReadPos++]; + if (iReadPos >= iBufSize) iReadPos = 0; + --len; + } + + // Fill remaining space with zero + while (len > 0) { + *p++ = 0; + --len; + } +} + +static void InitSDL() { + if (SDL_WasInit(SDL_INIT_EVERYTHING)) { + SDL_InitSubSystem(SDL_INIT_AUDIO); + } else { + SDL_Init(SDL_INIT_AUDIO | SDL_INIT_NOPARACHUTE); + } +} + +static void DestroySDL() { + if (SDL_WasInit(SDL_INIT_EVERYTHING & ~SDL_INIT_AUDIO)) { + SDL_QuitSubSystem(SDL_INIT_AUDIO); + } else { + SDL_Quit(); + } +} + +void SetupSound(void) { + SDL_AudioSpec spec; + + if (pSndBuffer != NULL) return; + + InitSDL(); + + spec.freq = 44100; + spec.format = AUDIO_S16SYS; + spec.channels = iDisStereo ? 1 : 2; + spec.samples = 512; + spec.callback = SOUND_FillAudio; + + if (SDL_OpenAudio(&spec, NULL) < 0) { + DestroySDL(); + return; + } + + iBufSize = BUFFER_SIZE; + if (iDisStereo) iBufSize /= 2; + + pSndBuffer = (short *)malloc(iBufSize * sizeof(short)); + if (pSndBuffer == NULL) { + SDL_CloseAudio(); + return; + } + + iReadPos = 0; + iWritePos = 0; + + SDL_PauseAudio(0); +} + +void RemoveSound(void) { + if (pSndBuffer == NULL) return; + + SDL_CloseAudio(); + DestroySDL(); + + free(pSndBuffer); + pSndBuffer = NULL; +} + +unsigned long SoundGetBytesBuffered(void) { + int size; + + if (pSndBuffer == NULL) return SOUNDSIZE; + + size = iReadPos - iWritePos; + if (size <= 0) size += iBufSize; + + if (size < iBufSize / 2) return SOUNDSIZE; + + return 0; +} + +void SoundFeedStreamData(unsigned char *pSound, long lBytes) { + short *p = (short *)pSound; + + if (pSndBuffer == NULL) return; + + while (lBytes > 0) { + if (((iWritePos + 1) % iBufSize) == iReadPos) break; + + pSndBuffer[iWritePos] = *p++; + + ++iWritePos; + if (iWritePos >= iBufSize) iWritePos = 0; + + lBytes -= sizeof(short); + } +} diff --git a/plugins/dfsound/spu.c b/plugins/dfsound/spu.c new file mode 100644 index 0000000..c086c06 --- /dev/null +++ b/plugins/dfsound/spu.c @@ -0,0 +1,1029 @@ +/*************************************************************************** + spu.c - description + ------------------- + begin : Wed May 15 2002 + copyright : (C) 2002 by Pete Bernert + email : BlackDove@addcom.de + ***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +#include "stdafx.h" + +#define _IN_SPU + +#include "externals.h" +#include "cfg.h" +#include "dsoundoss.h" +#include "regs.h" + +#ifdef ENABLE_NLS +#include <libintl.h> +#include <locale.h> +#define _(x) gettext(x) +#define N_(x) (x) +#else +#define _(x) (x) +#define N_(x) (x) +#endif + +#if defined (USEMACOSX) +static char * libraryName = N_("Mac OS X Sound"); +#elif defined (USEALSA) +static char * libraryName = N_("ALSA Sound"); +#elif defined (USEOSS) +static char * libraryName = N_("OSS Sound"); +#elif defined (USESDL) +static char * libraryName = N_("SDL Sound"); +#elif defined (USEPULSEAUDIO) +static char * libraryName = N_("PulseAudio Sound"); +#else +static char * libraryName = N_("NULL Sound"); +#endif + +static char * libraryInfo = N_("P.E.Op.S. Sound Driver V1.7\nCoded by Pete Bernert and the P.E.Op.S. team\n"); + +// globals + +// psx buffer / addresses + +unsigned short regArea[10000]; +unsigned short spuMem[256*1024]; +unsigned char * spuMemC; +unsigned char * pSpuIrq=0; +unsigned char * pSpuBuffer; +unsigned char * pMixIrq=0; + +// user settings + +int iVolume=3; +int iXAPitch=1; +int iUseTimer=2; +int iSPUIRQWait=1; +int iDebugMode=0; +int iRecordMode=0; +int iUseReverb=2; +int iUseInterpolation=2; +int iDisStereo=0; + +// MAIN infos struct for each channel + +SPUCHAN s_chan[MAXCHAN+1]; // channel + 1 infos (1 is security for fmod handling) +REVERBInfo rvb; + +unsigned long dwNoiseVal=1; // global noise generator +int iSpuAsyncWait=0; + +unsigned short spuCtrl=0; // some vars to store psx reg infos +unsigned short spuStat=0; +unsigned short spuIrq=0; +unsigned long spuAddr=0xffffffff; // address into spu mem +int bEndThread=0; // thread handlers +int bThreadEnded=0; +int bSpuInit=0; +int bSPUIsOpen=0; + +static pthread_t thread = (pthread_t)-1; // thread id (linux) + +unsigned long dwNewChannel=0; // flags for faster testing, if new channel starts + +void (CALLBACK *irqCallback)(void)=0; // func of main emu, called on spu irq +void (CALLBACK *cddavCallback)(unsigned short,unsigned short)=0; + +// certain globals (were local before, but with the new timeproc I need em global) + +static const int f[5][2] = { { 0, 0 }, + { 60, 0 }, + { 115, -52 }, + { 98, -55 }, + { 122, -60 } }; +int SSumR[NSSIZE]; +int SSumL[NSSIZE]; +int iFMod[NSSIZE]; +int iCycle = 0; +short * pS; + +int lastch=-1; // last channel processed on spu irq in timer mode +static int lastns=0; // last ns pos +static int iSecureStart=0; // secure start counter + +//////////////////////////////////////////////////////////////////////// +// CODE AREA +//////////////////////////////////////////////////////////////////////// + +// dirty inline func includes + +#include "reverb.c" +#include "adsr.c" + +//////////////////////////////////////////////////////////////////////// +// helpers for simple interpolation + +// +// easy interpolation on upsampling, no special filter, just "Pete's common sense" tm +// +// instead of having n equal sample values in a row like: +// ____ +// |____ +// +// we compare the current delta change with the next delta change. +// +// if curr_delta is positive, +// +// - and next delta is smaller (or changing direction): +// \. +// -__ +// +// - and next delta significant (at least twice) bigger: +// --_ +// \. +// +// - and next delta is nearly same: +// \. +// \. +// +// +// if curr_delta is negative, +// +// - and next delta is smaller (or changing direction): +// _-- +// / +// +// - and next delta significant (at least twice) bigger: +// / +// __- +// +// - and next delta is nearly same: +// / +// / +// + + +INLINE void InterpolateUp(int ch) +{ + if(s_chan[ch].SB[32]==1) // flag == 1? calc step and set flag... and don't change the value in this pass + { + const int id1=s_chan[ch].SB[30]-s_chan[ch].SB[29]; // curr delta to next val + const int id2=s_chan[ch].SB[31]-s_chan[ch].SB[30]; // and next delta to next-next val :) + + s_chan[ch].SB[32]=0; + + if(id1>0) // curr delta positive + { + if(id2<id1) + {s_chan[ch].SB[28]=id1;s_chan[ch].SB[32]=2;} + else + if(id2<(id1<<1)) + s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x10000L; + else + s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x20000L; + } + else // curr delta negative + { + if(id2>id1) + {s_chan[ch].SB[28]=id1;s_chan[ch].SB[32]=2;} + else + if(id2>(id1<<1)) + s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x10000L; + else + s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x20000L; + } + } + else + if(s_chan[ch].SB[32]==2) // flag 1: calc step and set flag... and don't change the value in this pass + { + s_chan[ch].SB[32]=0; + + s_chan[ch].SB[28]=(s_chan[ch].SB[28]*s_chan[ch].sinc)/0x20000L; + if(s_chan[ch].sinc<=0x8000) + s_chan[ch].SB[29]=s_chan[ch].SB[30]-(s_chan[ch].SB[28]*((0x10000/s_chan[ch].sinc)-1)); + else s_chan[ch].SB[29]+=s_chan[ch].SB[28]; + } + else // no flags? add bigger val (if possible), calc smaller step, set flag1 + s_chan[ch].SB[29]+=s_chan[ch].SB[28]; +} + +// +// even easier interpolation on downsampling, also no special filter, again just "Pete's common sense" tm +// + +INLINE void InterpolateDown(int ch) +{ + if(s_chan[ch].sinc>=0x20000L) // we would skip at least one val? + { + s_chan[ch].SB[29]+=(s_chan[ch].SB[30]-s_chan[ch].SB[29])/2; // add easy weight + if(s_chan[ch].sinc>=0x30000L) // we would skip even more vals? + s_chan[ch].SB[29]+=(s_chan[ch].SB[31]-s_chan[ch].SB[30])/2;// add additional next weight + } +} + +//////////////////////////////////////////////////////////////////////// +// helpers for gauss interpolation + +#define gval0 (((short*)(&s_chan[ch].SB[29]))[gpos]) +#define gval(x) (((short*)(&s_chan[ch].SB[29]))[(gpos+x)&3]) + +#include "gauss_i.h" + +//////////////////////////////////////////////////////////////////////// + +#include "xa.c" + +//////////////////////////////////////////////////////////////////////// +// START SOUND... called by main thread to setup a new sound on a channel +//////////////////////////////////////////////////////////////////////// + +INLINE void StartSound(int ch) +{ + StartADSR(ch); + StartREVERB(ch); + + s_chan[ch].pCurr=s_chan[ch].pStart; // set sample start + + s_chan[ch].s_1=0; // init mixing vars + s_chan[ch].s_2=0; + s_chan[ch].iSBPos=28; + + s_chan[ch].bNew=0; // init channel flags + s_chan[ch].bStop=0; + s_chan[ch].bOn=1; + + s_chan[ch].SB[29]=0; // init our interpolation helpers + s_chan[ch].SB[30]=0; + + if(iUseInterpolation>=2) // gauss interpolation? + {s_chan[ch].spos=0x30000L;s_chan[ch].SB[28]=0;} // -> start with more decoding + else {s_chan[ch].spos=0x10000L;s_chan[ch].SB[31]=0;} // -> no/simple interpolation starts with one 44100 decoding + + dwNewChannel&=~(1<<ch); // clear new channel bit +} + +//////////////////////////////////////////////////////////////////////// +// ALL KIND OF HELPERS +//////////////////////////////////////////////////////////////////////// + +INLINE void VoiceChangeFrequency(int ch) +{ + s_chan[ch].iUsedFreq=s_chan[ch].iActFreq; // -> take it and calc steps + s_chan[ch].sinc=s_chan[ch].iRawPitch<<4; + if(!s_chan[ch].sinc) s_chan[ch].sinc=1; + if(iUseInterpolation==1) s_chan[ch].SB[32]=1; // -> freq change in simle imterpolation mode: set flag +} + +//////////////////////////////////////////////////////////////////////// + +INLINE void FModChangeFrequency(int ch,int ns) +{ + int NP=s_chan[ch].iRawPitch; + + NP=((32768L+iFMod[ns])*NP)/32768L; + + if(NP>0x3fff) NP=0x3fff; + if(NP<0x1) NP=0x1; + + NP=(44100L*NP)/(4096L); // calc frequency + + s_chan[ch].iActFreq=NP; + s_chan[ch].iUsedFreq=NP; + s_chan[ch].sinc=(((NP/10)<<16)/4410); + if(!s_chan[ch].sinc) s_chan[ch].sinc=1; + if(iUseInterpolation==1) // freq change in simple interpolation mode + s_chan[ch].SB[32]=1; + iFMod[ns]=0; +} + +//////////////////////////////////////////////////////////////////////// + +// noise handler... just produces some noise data +// surely wrong... and no noise frequency (spuCtrl&0x3f00) will be used... +// and sometimes the noise will be used as fmod modulation... pfff + +INLINE int iGetNoiseVal(int ch) +{ + int fa; + + if((dwNoiseVal<<=1)&0x80000000L) + { + dwNoiseVal^=0x0040001L; + fa=((dwNoiseVal>>2)&0x7fff); + fa=-fa; + } + else fa=(dwNoiseVal>>2)&0x7fff; + + // mmm... depending on the noise freq we allow bigger/smaller changes to the previous val + fa=s_chan[ch].iOldNoise+((fa-s_chan[ch].iOldNoise)/((0x001f-((spuCtrl&0x3f00)>>9))+1)); + if(fa>32767L) fa=32767L; + if(fa<-32767L) fa=-32767L; + s_chan[ch].iOldNoise=fa; + + if(iUseInterpolation<2) // no gauss/cubic interpolation? + s_chan[ch].SB[29] = fa; // -> store noise val in "current sample" slot + return fa; +} + +//////////////////////////////////////////////////////////////////////// + +INLINE void StoreInterpolationVal(int ch,int fa) +{ + if(s_chan[ch].bFMod==2) // fmod freq channel + s_chan[ch].SB[29]=fa; + else + { + if((spuCtrl&0x4000)==0) fa=0; // muted? + else // else adjust + { + if(fa>32767L) fa=32767L; + if(fa<-32767L) fa=-32767L; + } + + if(iUseInterpolation>=2) // gauss/cubic interpolation + { + int gpos = s_chan[ch].SB[28]; + gval0 = fa; + gpos = (gpos+1) & 3; + s_chan[ch].SB[28] = gpos; + } + else + if(iUseInterpolation==1) // simple interpolation + { + s_chan[ch].SB[28] = 0; + s_chan[ch].SB[29] = s_chan[ch].SB[30]; // -> helpers for simple linear interpolation: delay real val for two slots, and calc the two deltas, for a 'look at the future behaviour' + s_chan[ch].SB[30] = s_chan[ch].SB[31]; + s_chan[ch].SB[31] = fa; + s_chan[ch].SB[32] = 1; // -> flag: calc new interolation + } + else s_chan[ch].SB[29]=fa; // no interpolation + } +} + +//////////////////////////////////////////////////////////////////////// + +INLINE int iGetInterpolationVal(int ch) +{ + int fa; + + if(s_chan[ch].bFMod==2) return s_chan[ch].SB[29]; + + switch(iUseInterpolation) + { + //--------------------------------------------------// + case 3: // cubic interpolation + { + long xd;int gpos; + xd = ((s_chan[ch].spos) >> 1)+1; + gpos = s_chan[ch].SB[28]; + + fa = gval(3) - 3*gval(2) + 3*gval(1) - gval0; + fa *= (xd - (2<<15)) / 6; + fa >>= 15; + fa += gval(2) - gval(1) - gval(1) + gval0; + fa *= (xd - (1<<15)) >> 1; + fa >>= 15; + fa += gval(1) - gval0; + fa *= xd; + fa >>= 15; + fa = fa + gval0; + + } break; + //--------------------------------------------------// + case 2: // gauss interpolation + { + int vl, vr;int gpos; + vl = (s_chan[ch].spos >> 6) & ~3; + gpos = s_chan[ch].SB[28]; + vr=(gauss[vl]*gval0)&~2047; + vr+=(gauss[vl+1]*gval(1))&~2047; + vr+=(gauss[vl+2]*gval(2))&~2047; + vr+=(gauss[vl+3]*gval(3))&~2047; + fa = vr>>11; + } break; + //--------------------------------------------------// + case 1: // simple interpolation + { + if(s_chan[ch].sinc<0x10000L) // -> upsampling? + InterpolateUp(ch); // --> interpolate up + else InterpolateDown(ch); // --> else down + fa=s_chan[ch].SB[29]; + } break; + //--------------------------------------------------// + default: // no interpolation + { + fa=s_chan[ch].SB[29]; + } break; + //--------------------------------------------------// + } + + return fa; +} + +//////////////////////////////////////////////////////////////////////// +// MAIN SPU FUNCTION +// here is the main job handler... thread, timer or direct func call +// basically the whole sound processing is done in this fat func! +//////////////////////////////////////////////////////////////////////// + +// 5 ms waiting phase, if buffer is full and no new sound has to get started +// .. can be made smaller (smallest val: 1 ms), but bigger waits give +// better performance + +#define PAUSE_W 5 +#define PAUSE_L 5000 + +//////////////////////////////////////////////////////////////////////// + +static void *MAINThread(void *arg) +{ + int s_1,s_2,fa,ns; +#ifndef _MACOSX + int voldiv = iVolume; +#else + const int voldiv = 2; +#endif + unsigned char * start;unsigned int nSample; + int ch,predict_nr,shift_factor,flags,d,s; + int bIRQReturn=0; + + while(!bEndThread) // until we are shutting down + { + // ok, at the beginning we are looking if there is + // enuff free place in the dsound/oss buffer to + // fill in new data, or if there is a new channel to start. + // if not, we wait (thread) or return (timer/spuasync) + // until enuff free place is available/a new channel gets + // started + + if(dwNewChannel) // new channel should start immedately? + { // (at least one bit 0 ... MAXCHANNEL is set?) + iSecureStart++; // -> set iSecure + if(iSecureStart>5) iSecureStart=0; // (if it is set 5 times - that means on 5 tries a new samples has been started - in a row, we will reset it, to give the sound update a chance) + } + else iSecureStart=0; // 0: no new channel should start + + while(!iSecureStart && !bEndThread && // no new start? no thread end? + (SoundGetBytesBuffered()>TESTSIZE)) // and still enuff data in sound buffer? + { + iSecureStart=0; // reset secure + + if(iUseTimer) return 0; // linux no-thread mode? bye + usleep(PAUSE_L); // else sleep for x ms (linux) + + if(dwNewChannel) iSecureStart=1; // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop + } + + //--------------------------------------------------// continue from irq handling in timer mode? + + if(lastch>=0) // will be -1 if no continue is pending + { + ch=lastch; ns=lastns; lastch=-1; // -> setup all kind of vars to continue + goto GOON; // -> directly jump to the continue point + } + + //--------------------------------------------------// + //- main channel loop -// + //--------------------------------------------------// + { + for(ch=0;ch<MAXCHAN;ch++) // loop em all... we will collect 1 ms of sound of each playing channel + { + if(s_chan[ch].bNew) StartSound(ch); // start new sound + if(!s_chan[ch].bOn) continue; // channel not playing? next + + if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq) // new psx frequency? + VoiceChangeFrequency(ch); + + ns=0; + while(ns<NSSIZE) // loop until 1 ms of data is reached + { + if(s_chan[ch].bFMod==1 && iFMod[ns]) // fmod freq channel + FModChangeFrequency(ch,ns); + + while(s_chan[ch].spos>=0x10000L) + { + if(s_chan[ch].iSBPos==28) // 28 reached? + { + start=s_chan[ch].pCurr; // set up the current pos + + if (start == (unsigned char*)-1) // special "stop" sign + { + s_chan[ch].bOn=0; // -> turn everything off + s_chan[ch].ADSRX.lVolume=0; + s_chan[ch].ADSRX.EnvelopeVol=0; + goto ENDX; // -> and done for this channel + } + + s_chan[ch].iSBPos=0; + + //////////////////////////////////////////// spu irq handler here? mmm... do it later + + s_1=s_chan[ch].s_1; + s_2=s_chan[ch].s_2; + + predict_nr=(int)*start;start++; + shift_factor=predict_nr&0xf; + predict_nr >>= 4; + flags=(int)*start;start++; + + // -------------------------------------- // + + for (nSample=0;nSample<28;start++) + { + d=(int)*start; + s=((d&0xf)<<12); + if(s&0x8000) s|=0xffff0000; + + fa=(s >> shift_factor); + fa=fa + ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6); + s_2=s_1;s_1=fa; + s=((d & 0xf0) << 8); + + s_chan[ch].SB[nSample++]=fa; + + if(s&0x8000) s|=0xffff0000; + fa=(s>>shift_factor); + fa=fa + ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6); + s_2=s_1;s_1=fa; + + s_chan[ch].SB[nSample++]=fa; + } + + //////////////////////////////////////////// irq check + + if(irqCallback && (spuCtrl&0x40)) // some callback and irq active? + { + if((pSpuIrq > start-16 && // irq address reached? + pSpuIrq <= start) || + ((flags&1) && // special: irq on looping addr, when stop/loop flag is set + (pSpuIrq > s_chan[ch].pLoop-16 && + pSpuIrq <= s_chan[ch].pLoop))) + { + s_chan[ch].iIrqDone=1; // -> debug flag + irqCallback(); // -> call main emu + + if(iSPUIRQWait) // -> option: wait after irq for main emu + { + iSpuAsyncWait=1; + bIRQReturn=1; + } + } + } + + //////////////////////////////////////////// flag handler + + if((flags&4) && (!s_chan[ch].bIgnoreLoop)) + s_chan[ch].pLoop=start-16; // loop adress + + if(flags&1) // 1: stop/loop + { + // We play this block out first... + //if(!(flags&2)) // 1+2: do loop... otherwise: stop + if(flags!=3 || s_chan[ch].pLoop==NULL) // PETE: if we don't check exactly for 3, loop hang ups will happen (DQ4, for example) + { // and checking if pLoop is set avoids crashes, yeah + start = (unsigned char*)-1; + } + else + { + start = s_chan[ch].pLoop; + } + } + + s_chan[ch].pCurr=start; // store values for next cycle + s_chan[ch].s_1=s_1; + s_chan[ch].s_2=s_2; + + if(bIRQReturn) // special return for "spu irq - wait for cpu action" + { + bIRQReturn=0; + if(iUseTimer!=2) + { + DWORD dwWatchTime=timeGetTime_spu()+2500; + + while(iSpuAsyncWait && !bEndThread && + timeGetTime_spu()<dwWatchTime) + usleep(1000L); + } + else + { + lastch=ch; + lastns=ns; + + return 0; + } + } + +GOON: ; + } + + fa=s_chan[ch].SB[s_chan[ch].iSBPos++]; // get sample data + + StoreInterpolationVal(ch,fa); // store val for later interpolation + + s_chan[ch].spos -= 0x10000L; + } + + if(s_chan[ch].bNoise) + fa=iGetNoiseVal(ch); // get noise val + else fa=iGetInterpolationVal(ch); // get sample val + + s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // mix adsr + + if(s_chan[ch].bFMod==2) // fmod freq channel + iFMod[ns]=s_chan[ch].sval; // -> store 1T sample data, use that to do fmod on next channel + else // no fmod freq channel + { + ////////////////////////////////////////////// + // ok, left/right sound volume (psx volume goes from 0 ... 0x3fff) + + if(s_chan[ch].iMute) + s_chan[ch].sval=0; // debug mute + else + { + SSumL[ns]+=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000L; + SSumR[ns]+=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000L; + } + + ////////////////////////////////////////////// + // now let us store sound data for reverb + + if(s_chan[ch].bRVBActive) StoreREVERB(ch,ns); + } + + //////////////////////////////////////////////// + // ok, go on until 1 ms data of this channel is collected + + ns++; + s_chan[ch].spos += s_chan[ch].sinc; + + } +ENDX: ; + } + } + + //---------------------------------------------------// + //- here we have another 1 ms of sound data + //---------------------------------------------------// + // mix XA infos (if any) + + MixXA(); + + /////////////////////////////////////////////////////// + // mix all channels (including reverb) into one buffer + + if(iDisStereo) // no stereo? + { + int dl, dr; + for (ns = 0; ns < NSSIZE; ns++) + { + SSumL[ns] += MixREVERBLeft(ns); + + dl = SSumL[ns] / voldiv; SSumL[ns] = 0; + if (dl < -32767) dl = -32767; if (dl > 32767) dl = 32767; + + SSumR[ns] += MixREVERBRight(); + + dr = SSumR[ns] / voldiv; SSumR[ns] = 0; + if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767; + *pS++ = (dl + dr) / 2; + } + } + else // stereo: + for (ns = 0; ns < NSSIZE; ns++) + { + SSumL[ns] += MixREVERBLeft(ns); + + d = SSumL[ns] / voldiv; SSumL[ns] = 0; + if (d < -32767) d = -32767; if (d > 32767) d = 32767; + *pS++ = d; + + SSumR[ns] += MixREVERBRight(); + + d = SSumR[ns] / voldiv; SSumR[ns] = 0; + if(d < -32767) d = -32767; if(d > 32767) d = 32767; + *pS++ = d; + } + + ////////////////////////////////////////////////////// + // special irq handling in the decode buffers (0x0000-0x1000) + // we know: + // the decode buffers are located in spu memory in the following way: + // 0x0000-0x03ff CD audio left + // 0x0400-0x07ff CD audio right + // 0x0800-0x0bff Voice 1 + // 0x0c00-0x0fff Voice 3 + // and decoded data is 16 bit for one sample + // we assume: + // even if voices 1/3 are off or no cd audio is playing, the internal + // play positions will move on and wrap after 0x400 bytes. + // Therefore: we just need a pointer from spumem+0 to spumem+3ff, and + // increase this pointer on each sample by 2 bytes. If this pointer + // (or 0x400 offsets of this pointer) hits the spuirq address, we generate + // an IRQ. Only problem: the "wait for cpu" option is kinda hard to do here + // in some of Peops timer modes. So: we ignore this option here (for now). + + if(pMixIrq && irqCallback) + { + for(ns=0;ns<NSSIZE;ns++) + { + if((spuCtrl&0x40) && pSpuIrq && pSpuIrq<spuMemC+0x1000) + { + for(ch=0;ch<4;ch++) + { + if(pSpuIrq>=pMixIrq+(ch*0x400) && pSpuIrq<pMixIrq+(ch*0x400)+2) + {irqCallback();s_chan[ch].iIrqDone=1;} + } + } + pMixIrq+=2;if(pMixIrq>spuMemC+0x3ff) pMixIrq=spuMemC; + } + } + + InitREVERB(); + + // feed the sound + // wanna have around 1/60 sec (16.666 ms) updates + if (iCycle++ > 16) + { + SoundFeedStreamData((unsigned char *)pSpuBuffer, + ((unsigned char *)pS) - ((unsigned char *)pSpuBuffer)); + pS = (short *)pSpuBuffer; + iCycle = 0; + } + } + + // end of big main loop... + + bThreadEnded = 1; + + return 0; +} + +// SPU ASYNC... even newer epsxe func +// 1 time every 'cycle' cycles... harhar + +void CALLBACK SPUasync(unsigned long cycle) +{ + if(iSpuAsyncWait) + { + iSpuAsyncWait++; + if(iSpuAsyncWait<=64) return; + iSpuAsyncWait=0; + } + + if(iUseTimer==2) // special mode, only used in Linux by this spu (or if you enable the experimental Windows mode) + { + if(!bSpuInit) return; // -> no init, no call + + MAINThread(0); // -> linux high-compat mode + } +} + +// SPU UPDATE... new epsxe func +// 1 time every 32 hsync lines +// (312/32)x50 in pal +// (262/32)x60 in ntsc + +// since epsxe 1.5.2 (linux) uses SPUupdate, not SPUasync, I will +// leave that func in the linux port, until epsxe linux is using +// the async function as well + +void CALLBACK SPUupdate(void) +{ + SPUasync(0); +} + +// XA AUDIO + +void CALLBACK SPUplayADPCMchannel(xa_decode_t *xap) +{ + if(!xap) return; + if(!xap->freq) return; // no xa freq ? bye + + FeedXA(xap); // call main XA feeder +} + +// CDDA AUDIO +void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes) +{ + if (!pcm) return; + if (nbytes<=0) return; + + FeedCDDA((unsigned char *)pcm, nbytes); +} + +// SETUPTIMER: init of certain buffers and threads/timers +void SetupTimer(void) +{ + memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers + memset(SSumL,0,NSSIZE*sizeof(int)); + memset(iFMod,0,NSSIZE*sizeof(int)); + pS=(short *)pSpuBuffer; // setup soundbuffer pointer + + bEndThread=0; // init thread vars + bThreadEnded=0; + bSpuInit=1; // flag: we are inited + + if(!iUseTimer) // linux: use thread + { + pthread_create(&thread, NULL, MAINThread, NULL); + } +} + +// REMOVETIMER: kill threads/timers +void RemoveTimer(void) +{ + bEndThread=1; // raise flag to end thread + + if(!iUseTimer) // linux tread? + { + int i=0; + while(!bThreadEnded && i<2000) {usleep(1000L);i++;} // -> wait until thread has ended + if(thread!=(pthread_t)-1) {pthread_cancel(thread);thread=(pthread_t)-1;} // -> cancel thread anyway + } + + bThreadEnded=0; // no more spu is running + bSpuInit=0; +} + +// SETUPSTREAMS: init most of the spu buffers +void SetupStreams(void) +{ + int i; + + pSpuBuffer=(unsigned char *)malloc(32768); // alloc mixing buffer + + if(iUseReverb==1) i=88200*2; + else i=NSSIZE*2; + + sRVBStart = (int *)malloc(i*4); // alloc reverb buffer + memset(sRVBStart,0,i*4); + sRVBEnd = sRVBStart + i; + sRVBPlay = sRVBStart; + + XAStart = // alloc xa buffer + (uint32_t *)malloc(44100 * sizeof(uint32_t)); + XAEnd = XAStart + 44100; + XAPlay = XAStart; + XAFeed = XAStart; + + CDDAStart = // alloc cdda buffer + (uint32_t *)malloc(16384 * sizeof(uint32_t)); + CDDAEnd = CDDAStart + 16384; + CDDAPlay = CDDAStart; + CDDAFeed = CDDAStart + 1; + + for(i=0;i<MAXCHAN;i++) // loop sound channels + { +// we don't use mutex sync... not needed, would only +// slow us down: +// s_chan[i].hMutex=CreateMutex(NULL,FALSE,NULL); + s_chan[i].ADSRX.SustainLevel = 1024; // -> init sustain + s_chan[i].iMute=0; + s_chan[i].iIrqDone=0; + s_chan[i].pLoop=spuMemC; + s_chan[i].pStart=spuMemC; + s_chan[i].pCurr=spuMemC; + } + + pMixIrq=spuMemC; // enable decoded buffer irqs by setting the address +} + +// REMOVESTREAMS: free most buffer +void RemoveStreams(void) +{ + free(pSpuBuffer); // free mixing buffer + pSpuBuffer = NULL; + free(sRVBStart); // free reverb buffer + sRVBStart = NULL; + free(XAStart); // free XA buffer + XAStart = NULL; + free(CDDAStart); // free CDDA buffer + CDDAStart = NULL; +} + +// INIT/EXIT STUFF + +// SPUINIT: this func will be called first by the main emu +long CALLBACK SPUinit(void) +{ + spuMemC = (unsigned char *)spuMem; // just small setup + memset((void *)&rvb, 0, sizeof(REVERBInfo)); + InitADSR(); + + iVolume = 3; + iReverbOff = -1; + spuIrq = 0; + spuAddr = 0xffffffff; + bEndThread = 0; + bThreadEnded = 0; + spuMemC = (unsigned char *)spuMem; + pMixIrq = 0; + memset((void *)s_chan, 0, (MAXCHAN + 1) * sizeof(SPUCHAN)); + pSpuIrq = 0; + iSPUIRQWait = 1; + lastch = -1; + + ReadConfig(); // read user stuff + SetupStreams(); // prepare streaming + + return 0; +} + +// SPUOPEN: called by main emu after init +long CALLBACK SPUopen(void) +{ + if (bSPUIsOpen) return 0; // security for some stupid main emus + + SetupSound(); // setup sound (before init!) + SetupTimer(); // timer for feeding data + + bSPUIsOpen = 1; + + return PSE_SPU_ERR_SUCCESS; +} + +// SPUCLOSE: called before shutdown +long CALLBACK SPUclose(void) +{ + if (!bSPUIsOpen) return 0; // some security + + bSPUIsOpen = 0; // no more open + + RemoveTimer(); // no more feeding + RemoveSound(); // no more sound handling + + return 0; +} + +// SPUSHUTDOWN: called by main emu on final exit +long CALLBACK SPUshutdown(void) +{ + SPUclose(); + RemoveStreams(); // no more streaming + + return 0; +} + +// SPUTEST: we don't test, we are always fine ;) +long CALLBACK SPUtest(void) +{ + return 0; +} + +// SPUCONFIGURE: call config dialog +long CALLBACK SPUconfigure(void) +{ +#ifdef _MACOSX + DoConfiguration(); +#else + StartCfgTool("CFG"); +#endif + return 0; +} + +// SPUABOUT: show about window +void CALLBACK SPUabout(void) +{ +#ifdef _MACOSX + DoAbout(); +#else + StartCfgTool("ABOUT"); +#endif +} + +// SETUP CALLBACKS +// this functions will be called once, +// passes a callback that should be called on SPU-IRQ/cdda volume change +void CALLBACK SPUregisterCallback(void (CALLBACK *callback)(void)) +{ + irqCallback = callback; +} + +void CALLBACK SPUregisterCDDAVolume(void (CALLBACK *CDDAVcallback)(unsigned short,unsigned short)) +{ + cddavCallback = CDDAVcallback; +} + +// COMMON PLUGIN INFO FUNCS +char * CALLBACK PSEgetLibName(void) +{ + return _(libraryName); +} + +unsigned long CALLBACK PSEgetLibType(void) +{ + return PSE_LT_SPU; +} + +unsigned long CALLBACK PSEgetLibVersion(void) +{ + return (1 << 16) | (6 << 8); +} + +char * SPUgetLibInfos(void) +{ + return _(libraryInfo); +} diff --git a/plugins/dfsound/spu.h b/plugins/dfsound/spu.h new file mode 100644 index 0000000..8912684 --- /dev/null +++ b/plugins/dfsound/spu.h @@ -0,0 +1,21 @@ +/***************************************************************************
+ spu.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+void SetupTimer(void);
+void RemoveTimer(void);
+void CALLBACK SPUplayADPCMchannel(xa_decode_t *xap);
+void CALLBACK SPUplayCDDAchannel(short *pcm, int bytes);
\ No newline at end of file diff --git a/plugins/dfsound/spucfg-0.1df/dfsound.glade2 b/plugins/dfsound/spucfg-0.1df/dfsound.glade2 new file mode 100644 index 0000000..c071a09 --- /dev/null +++ b/plugins/dfsound/spucfg-0.1df/dfsound.glade2 @@ -0,0 +1,308 @@ +<?xml version="1.0"?> +<glade-interface> + <!-- interface-requires gtk+ 2.8 --> + <!-- interface-naming-policy project-wide --> + <widget class="GtkWindow" id="CfgWnd"> + <property name="visible">True</property> + <property name="border_width">10</property> + <property name="title" translatable="yes">Configure Sound</property> + <property name="resizable">False</property> + <property name="modal">True</property> + <property name="window_position">center</property> + <property name="type_hint">dialog</property> + <signal name="destroy" handler="on_CfgWnd_destroy"/> + <child> + <widget class="GtkVBox" id="vbox1"> + <property name="visible">True</property> + <property name="orientation">vertical</property> + <property name="spacing">6</property> + <child> + <widget class="GtkFrame" id="frame1"> + <property name="visible">True</property> + <property name="label_xalign">0</property> + <child> + <widget class="GtkAlignment" id="alignment4"> + <property name="visible">True</property> + <property name="top_padding">6</property> + <property name="bottom_padding">6</property> + <property name="left_padding">12</property> + <property name="right_padding">12</property> + <child> + <widget class="GtkTable" id="table1"> + <property name="visible">True</property> + <property name="border_width">6</property> + <property name="n_rows">3</property> + <property name="n_columns">2</property> + <property name="column_spacing">6</property> + <property name="row_spacing">6</property> + <child> + <widget class="GtkLabel" id="label1v"> + <property name="visible">True</property> + <property name="xalign">0</property> + <property name="label" translatable="yes">Volume:</property> + <property name="justify">right</property> + </widget> + <packing> + <property name="x_options">GTK_FILL</property> + <property name="y_options"></property> + </packing> + </child> + <child> + <widget class="GtkLabel" id="label9"> + <property name="visible">True</property> + <property name="xalign">0</property> + <property name="label" translatable="yes">Interpolation:</property> + <property name="justify">right</property> + </widget> + <packing> + <property name="top_attach">2</property> + <property name="bottom_attach">3</property> + <property name="x_options">GTK_FILL</property> + <property name="y_options"></property> + </packing> + </child> + <child> + <widget class="GtkLabel" id="label8"> + <property name="visible">True</property> + <property name="xalign">0</property> + <property name="label" translatable="yes">Reverb:</property> + <property name="justify">right</property> + </widget> + <packing> + <property name="top_attach">1</property> + <property name="bottom_attach">2</property> + <property name="x_options">GTK_FILL</property> + <property name="y_options"></property> + </packing> + </child> + <child> + <widget class="GtkComboBox" id="cbVolume2"> + <property name="visible">True</property> + <property name="items" translatable="yes">Low +Medium +Loud +Loudest</property> + </widget> + <packing> + <property name="left_attach">1</property> + <property name="right_attach">2</property> + <property name="y_options">GTK_FILL</property> + </packing> + </child> + <child> + <widget class="GtkComboBox" id="cbReverb2"> + <property name="visible">True</property> + <property name="items" translatable="yes">Off +Simple +Playstation</property> + </widget> + <packing> + <property name="left_attach">1</property> + <property name="right_attach">2</property> + <property name="top_attach">1</property> + <property name="bottom_attach">2</property> + <property name="x_options">GTK_FILL</property> + <property name="y_options">GTK_FILL</property> + </packing> + </child> + <child> + <widget class="GtkComboBox" id="cbInterpolation2"> + <property name="visible">True</property> + <property name="items" translatable="yes">None +Simple +Gaussian +Cubic</property> + </widget> + <packing> + <property name="left_attach">1</property> + <property name="right_attach">2</property> + <property name="top_attach">2</property> + <property name="bottom_attach">3</property> + <property name="x_options">GTK_FILL</property> + <property name="y_options">GTK_FILL</property> + </packing> + </child> + </widget> + </child> + </widget> + </child> + <child> + <widget class="GtkLabel" id="label10"> + <property name="visible">True</property> + <property name="label" translatable="yes"><b>General</b></property> + <property name="use_markup">True</property> + </widget> + <packing> + <property name="type">label_item</property> + </packing> + </child> + </widget> + <packing> + <property name="position">0</property> + </packing> + </child> + <child> + <widget class="GtkFrame" id="frame2"> + <property name="visible">True</property> + <property name="label_xalign">0</property> + <child> + <widget class="GtkAlignment" id="alignment2"> + <property name="visible">True</property> + <property name="top_padding">6</property> + <property name="bottom_padding">6</property> + <property name="left_padding">12</property> + <property name="right_padding">12</property> + <child> + <widget class="GtkVBox" id="vbox3"> + <property name="visible">True</property> + <property name="border_width">6</property> + <property name="orientation">vertical</property> + <property name="spacing">6</property> + <child> + <widget class="GtkCheckButton" id="chkXASpeed"> + <property name="label" translatable="yes">Adjust XA speed</property> + <property name="visible">True</property> + <property name="can_focus">True</property> + <property name="receives_default">False</property> + <property name="tooltip" translatable="yes">Choose this if XA music is played too quickly.</property> + <property name="use_underline">True</property> + <property name="draw_indicator">True</property> + </widget> + <packing> + <property name="expand">False</property> + <property name="fill">False</property> + <property name="position">0</property> + </packing> + </child> + </widget> + </child> + </widget> + </child> + <child> + <widget class="GtkLabel" id="label11"> + <property name="visible">True</property> + <property name="label" translatable="yes"><b>XA Music</b></property> + <property name="use_markup">True</property> + </widget> + <packing> + <property name="type">label_item</property> + </packing> + </child> + </widget> + <packing> + <property name="position">1</property> + </packing> + </child> + <child> + <widget class="GtkFrame" id="frame3"> + <property name="visible">True</property> + <property name="label_xalign">0</property> + <child> + <widget class="GtkAlignment" id="alignment3"> + <property name="visible">True</property> + <property name="top_padding">6</property> + <property name="bottom_padding">6</property> + <property name="left_padding">12</property> + <property name="right_padding">12</property> + <child> + <widget class="GtkVBox" id="vbox4"> + <property name="visible">True</property> + <property name="border_width">6</property> + <property name="orientation">vertical</property> + <property name="spacing">6</property> + <child> + <widget class="GtkCheckButton" id="chkHiCompat"> + <property name="label" translatable="yes">High compatibility mode</property> + <property name="visible">True</property> + <property name="can_focus">True</property> + <property name="receives_default">False</property> + <property name="tooltip" translatable="yes">Use the asynchronous SPU interface.</property> + <property name="use_underline">True</property> + <property name="draw_indicator">True</property> + </widget> + <packing> + <property name="expand">False</property> + <property name="fill">False</property> + <property name="position">0</property> + </packing> + </child> + <child> + <widget class="GtkCheckButton" id="chkIRQWait"> + <property name="label" translatable="yes">SPU IRQ Wait</property> + <property name="visible">True</property> + <property name="can_focus">True</property> + <property name="receives_default">False</property> + <property name="tooltip" translatable="yes">Wait for CPU; only useful for some games.</property> + <property name="use_underline">True</property> + <property name="draw_indicator">True</property> + </widget> + <packing> + <property name="expand">False</property> + <property name="fill">False</property> + <property name="position">1</property> + </packing> + </child> + <child> + <widget class="GtkCheckButton" id="chkDisStereo"> + <property name="label" translatable="yes">Single channel sound</property> + <property name="visible">True</property> + <property name="can_focus">True</property> + <property name="receives_default">False</property> + <property name="tooltip" translatable="yes">Play only one channel for a performance boost.</property> + <property name="use_underline">True</property> + <property name="draw_indicator">True</property> + </widget> + <packing> + <property name="expand">False</property> + <property name="fill">False</property> + <property name="position">2</property> + </packing> + </child> + </widget> + </child> + </widget> + </child> + <child> + <widget class="GtkLabel" id="label12"> + <property name="visible">True</property> + <property name="label" translatable="yes"><b>Compatibility</b></property> + <property name="use_markup">True</property> + </widget> + <packing> + <property name="type">label_item</property> + </packing> + </child> + </widget> + <packing> + <property name="position">2</property> + </packing> + </child> + <child> + <widget class="GtkHButtonBox" id="hbuttonbox1"> + <property name="visible">True</property> + <property name="spacing">12</property> + <property name="layout_style">end</property> + <child> + <widget class="GtkButton" id="btn_close"> + <property name="label">gtk-close</property> + <property name="visible">True</property> + <property name="can_focus">True</property> + <property name="can_default">True</property> + <property name="receives_default">False</property> + <property name="use_stock">True</property> + </widget> + <packing> + <property name="expand">False</property> + <property name="fill">False</property> + <property name="position">0</property> + </packing> + </child> + </widget> + <packing> + <property name="position">3</property> + </packing> + </child> + </widget> + </child> + </widget> +</glade-interface> diff --git a/plugins/dfsound/spucfg-0.1df/main.c b/plugins/dfsound/spucfg-0.1df/main.c new file mode 100644 index 0000000..dd38645 --- /dev/null +++ b/plugins/dfsound/spucfg-0.1df/main.c @@ -0,0 +1,258 @@ +#include "config.h" + +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <sys/stat.h> + +#include <glade/glade.h> +#include <gtk/gtk.h> + +#ifdef ENABLE_NLS +#include <libintl.h> +#include <locale.h> +#endif + +#define READBINARY "rb" +#define WRITEBINARY "wb" +#define CONFIG_FILENAME "dfsound.cfg" + +void SaveConfig(GtkWidget *widget, gpointer user_datal); + +/* This function checks for the value being outside the accepted range, + and returns the appropriate boundary value */ +int set_limit (char *p, int len, int lower, int upper) +{ + int val = 0; + + if (p) + val = atoi(p + len); + + if (val < lower) + val = lower; + if (val > upper) + val = upper; + + return val; +} + +void on_about_clicked (GtkWidget *widget, gpointer user_data) +{ + gtk_widget_destroy (widget); + exit (0); +} + +void OnConfigClose(GtkWidget *widget, gpointer user_data) +{ + GladeXML *xml = (GladeXML *)user_data; + + gtk_widget_destroy(glade_xml_get_widget(xml, "CfgWnd")); + gtk_exit(0); +} + +int main(int argc, char *argv[]) +{ + GtkWidget *widget; + GladeXML *xml; + FILE *in; + char t[256]; + int len, val = 0; + char *pB, *p; + char cfg[255]; + +#ifdef ENABLE_NLS + setlocale (LC_ALL, ""); + bindtextdomain (GETTEXT_PACKAGE, LOCALE_DIR); + bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8"); + textdomain (GETTEXT_PACKAGE); +#endif + + if (argc != 2) { + printf ("Usage: cfgDFSound {ABOUT | CFG}\n"); + return 0; + } + + if (strcmp(argv[1], "CFG") != 0 && strcmp(argv[1], "ABOUT") != 0) { + printf ("Usage: cfgDFSound {ABOUT | CFG}\n"); + return 0; + } + + gtk_set_locale(); + gtk_init(&argc, &argv); + + if (strcmp(argv[1], "ABOUT") == 0) { + const char *authors[]= {"Pete Bernert and the P.E.Op.S. team", "Ryan Schultz", "Andrew Burton", NULL}; + widget = gtk_about_dialog_new (); + gtk_about_dialog_set_name (GTK_ABOUT_DIALOG (widget), "dfsound PCSX Sound Plugin"); + gtk_about_dialog_set_version (GTK_ABOUT_DIALOG (widget), "1.6"); + gtk_about_dialog_set_authors (GTK_ABOUT_DIALOG (widget), authors); + gtk_about_dialog_set_website (GTK_ABOUT_DIALOG (widget), "http://pcsx-df.sourceforge.net/"); + + g_signal_connect_data(GTK_OBJECT(widget), "response", + GTK_SIGNAL_FUNC(on_about_clicked), NULL, NULL, G_CONNECT_AFTER); + + gtk_widget_show (widget); + gtk_main(); + + return 0; + } + + xml = glade_xml_new(DATADIR "dfsound.glade2", "CfgWnd", NULL); + if (!xml) { + g_warning("We could not load the interface!"); + return 255; + } + + strcpy(cfg, CONFIG_FILENAME); + + in = fopen(cfg, READBINARY); + if (in) { + pB = (char *)malloc(32767); + memset(pB, 0, 32767); + len = fread(pB, 1, 32767, in); + fclose(in); + } else { + pB = 0; + printf ("Error - no configuration file\n"); + /* TODO Raise error - no configuration file */ + } + + /* ADB TODO Replace a lot of the following with common functions */ + if (pB) { + strcpy(t, "\nVolume"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + val = set_limit (p, len, 0, 4); + } else val = 2; + + gtk_combo_box_set_active(GTK_COMBO_BOX (glade_xml_get_widget(xml, "cbVolume2")), val); + + if (pB) { + strcpy(t, "\nUseInterpolation"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + val = set_limit (p, len, 0, 3); + } else val = 2; + + gtk_combo_box_set_active(GTK_COMBO_BOX (glade_xml_get_widget(xml, "cbInterpolation2")), val); + + if (pB) { + strcpy(t, "\nXAPitch"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + val = set_limit (p, len, 0, 1); + } else val = 0; + + gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON (glade_xml_get_widget(xml, "chkXASpeed")), val); + + if (pB) { + strcpy(t, "\nHighCompMode"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + val = set_limit (p, len, 0, 1); + } else val = 0; + + gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON (glade_xml_get_widget(xml, "chkHiCompat")), val); + + if (pB) { + strcpy(t, "\nSPUIRQWait"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + + val = set_limit (p, len, 0, 1); + } else val = 1; + + gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON (glade_xml_get_widget(xml, "chkIRQWait")), val); + + if (pB) { + strcpy(t, "\nDisStereo"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + + val = set_limit (p, len, 0, 1); + } else val = 0; + + gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON (glade_xml_get_widget(xml, "chkDisStereo")), val); + + if (pB) { + strcpy(t, "\nUseReverb"); + p = strstr(pB, t); + if (p) { + p = strstr(p, "="); + len = 1; + } + val = set_limit (p, len, 0, 2); + } else val = 2; + + gtk_combo_box_set_active(GTK_COMBO_BOX(glade_xml_get_widget(xml, "cbReverb2")), val); + + if (pB) + free(pB); + + widget = glade_xml_get_widget(xml, "CfgWnd"); + g_signal_connect_data(GTK_OBJECT(widget), "destroy", + GTK_SIGNAL_FUNC(SaveConfig), xml, NULL, 0); + + widget = glade_xml_get_widget(xml, "btn_close"); + g_signal_connect_data(GTK_OBJECT(widget), "clicked", + GTK_SIGNAL_FUNC(OnConfigClose), xml, NULL, G_CONNECT_AFTER); + + gtk_main(); + return 0; +} + +void SaveConfig(GtkWidget *widget, gpointer user_data) +{ + GladeXML *xml = (GladeXML *)user_data; + FILE *fp; + int val; + + fp = fopen(CONFIG_FILENAME, WRITEBINARY); + if (fp == NULL) { + fprintf(stderr, "Unable to write to configuration file %s!\n", CONFIG_FILENAME); + gtk_exit(0); + } + + val = gtk_combo_box_get_active(GTK_COMBO_BOX(glade_xml_get_widget(xml, "cbVolume2"))); + fprintf(fp, "\nVolume = %d\n", val); + + val = gtk_combo_box_get_active(GTK_COMBO_BOX(glade_xml_get_widget(xml, "cbInterpolation2"))); + fprintf(fp, "\nUseInterpolation = %d\n", val); + + val = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(glade_xml_get_widget(xml, "chkXASpeed"))); + fprintf(fp, "\nXAPitch = %d\n", val); + + val = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(glade_xml_get_widget(xml, "chkHiCompat"))); + fprintf(fp, "\nHighCompMode = %d\n", val); + + val = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(glade_xml_get_widget(xml, "chkIRQWait"))); + fprintf(fp, "\nSPUIRQWait = %d\n", val); + + val = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(glade_xml_get_widget(xml, "chkDisStereo"))); + fprintf(fp, "\nDisStereo = %d\n", val); + + val = gtk_combo_box_get_active(GTK_COMBO_BOX(glade_xml_get_widget(xml, "cbReverb2"))); + fprintf(fp, "\nUseReverb = %d\n", val); + + fclose(fp); + gtk_exit(0); +} diff --git a/plugins/dfsound/stdafx.h b/plugins/dfsound/stdafx.h new file mode 100644 index 0000000..8be8848 --- /dev/null +++ b/plugins/dfsound/stdafx.h @@ -0,0 +1,46 @@ +/*************************************************************************** + StdAfx.h - description + ------------------- + begin : Wed May 15 2002 + copyright : (C) 2002 by Pete Bernert + email : BlackDove@addcom.de + ***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +#ifndef _MACOSX +#include "config.h" +#endif +#include <stdio.h> +#include <stdlib.h> +#include <sys/ioctl.h> +#include <unistd.h> +#include <fcntl.h> +#ifdef USEOSS +#include <sys/soundcard.h> +#endif +#include <unistd.h> +#include <pthread.h> +#define RRand(range) (random()%range) +#include <string.h> +#include <sys/time.h> +#include <math.h> + +#undef CALLBACK +#define CALLBACK +#define DWORD unsigned long +#define LOWORD(l) ((unsigned short)(l)) +#define HIWORD(l) ((unsigned short)(((unsigned long)(l) >> 16) & 0xFFFF)) + +#ifndef INLINE +#define INLINE inline +#endif + +#include "psemuxa.h" diff --git a/plugins/dfsound/xa.c b/plugins/dfsound/xa.c new file mode 100644 index 0000000..fdae4f9 --- /dev/null +++ b/plugins/dfsound/xa.c @@ -0,0 +1,410 @@ +/*************************************************************************** + xa.c - description + ------------------- + begin : Wed May 15 2002 + copyright : (C) 2002 by Pete Bernert + email : BlackDove@addcom.de + ***************************************************************************/ +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. See also the license.txt file for * + * additional informations. * + * * + ***************************************************************************/ + +#include "stdafx.h" +#define _IN_XA +#include <stdint.h> + +// will be included from spu.c +#ifdef _IN_SPU + +//////////////////////////////////////////////////////////////////////// +// XA GLOBALS +//////////////////////////////////////////////////////////////////////// + +xa_decode_t * xapGlobal=0; + +uint32_t * XAFeed = NULL; +uint32_t * XAPlay = NULL; +uint32_t * XAStart = NULL; +uint32_t * XAEnd = NULL; + +uint32_t XARepeat = 0; +uint32_t XALastVal = 0; + +uint32_t * CDDAFeed = NULL; +uint32_t * CDDAPlay = NULL; +uint32_t * CDDAStart = NULL; +uint32_t * CDDAEnd = NULL; + +int iLeftXAVol = 32767; +int iRightXAVol = 32767; + +static int gauss_ptr = 0; +static int gauss_window[8] = {0, 0, 0, 0, 0, 0, 0, 0}; + +#define gvall0 gauss_window[gauss_ptr] +#define gvall(x) gauss_window[(gauss_ptr+x)&3] +#define gvalr0 gauss_window[4+gauss_ptr] +#define gvalr(x) gauss_window[4+((gauss_ptr+x)&3)] + +//////////////////////////////////////////////////////////////////////// +// MIX XA & CDDA +//////////////////////////////////////////////////////////////////////// + +INLINE void MixXA(void) +{ + int ns; + uint32_t l; + + for(ns=0;ns<NSSIZE && XAPlay!=XAFeed;ns++) + { + XALastVal=*XAPlay++; + if(XAPlay==XAEnd) XAPlay=XAStart; +#ifdef XA_HACK + SSumL[ns]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32768; + SSumR[ns]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32768; +#else + SSumL[ns]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32767; + SSumR[ns]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32767; +#endif + } + + if(XAPlay==XAFeed && XARepeat) + { + XARepeat--; + for(;ns<NSSIZE;ns++) + { +#ifdef XA_HACK + SSumL[ns]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32768; + SSumR[ns]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32768; +#else + SSumL[ns]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32767; + SSumR[ns]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32767; +#endif + } + } + + for(ns=0;ns<NSSIZE && CDDAPlay!=CDDAFeed && (CDDAPlay!=CDDAEnd-1||CDDAFeed!=CDDAStart);ns++) + { + l=*CDDAPlay++; + if(CDDAPlay==CDDAEnd) CDDAPlay=CDDAStart; + SSumL[ns]+=(((short)(l&0xffff)) * iLeftXAVol)/32767; + SSumR[ns]+=(((short)((l>>16)&0xffff)) * iRightXAVol)/32767; + } +} + +//////////////////////////////////////////////////////////////////////// +// small linux time helper... only used for watchdog +//////////////////////////////////////////////////////////////////////// + +unsigned long timeGetTime_spu() +{ + struct timeval tv; + gettimeofday(&tv, 0); // well, maybe there are better ways + return tv.tv_sec * 1000 + tv.tv_usec/1000; // to do that, but at least it works +} + +//////////////////////////////////////////////////////////////////////// +// FEED XA +//////////////////////////////////////////////////////////////////////// + +INLINE void FeedXA(xa_decode_t *xap) +{ + int sinc,spos,i,iSize,iPlace,vl,vr; + + if(!bSPUIsOpen) return; + + xapGlobal = xap; // store info for save states + XARepeat = 100; // set up repeat + +#ifdef XA_HACK + iSize=((45500*xap->nsamples)/xap->freq); // get size +#else + iSize=((44100*xap->nsamples)/xap->freq); // get size +#endif + if(!iSize) return; // none? bye + + if(XAFeed<XAPlay) iPlace=XAPlay-XAFeed; // how much space in my buf? + else iPlace=(XAEnd-XAFeed) + (XAPlay-XAStart); + + if(iPlace==0) return; // no place at all + + //----------------------------------------------------// + if(iXAPitch) // pitch change option? + { + static DWORD dwLT=0; + static DWORD dwFPS=0; + static int iFPSCnt=0; + static int iLastSize=0; + static DWORD dwL1=0; + DWORD dw=timeGetTime_spu(),dw1,dw2; + + iPlace=iSize; + + dwFPS+=dw-dwLT;iFPSCnt++; + + dwLT=dw; + + if(iFPSCnt>=10) + { + if(!dwFPS) dwFPS=1; + dw1=1000000/dwFPS; + if(dw1>=(dwL1-100) && dw1<=(dwL1+100)) dw1=dwL1; + else dwL1=dw1; + dw2=(xap->freq*100/xap->nsamples); + if((!dw1)||((dw2+100)>=dw1)) iLastSize=0; + else + { + iLastSize=iSize*dw2/dw1; + if(iLastSize>iPlace) iLastSize=iPlace; + iSize=iLastSize; + } + iFPSCnt=0;dwFPS=0; + } + else + { + if(iLastSize) iSize=iLastSize; + } + } + //----------------------------------------------------// + + spos=0x10000L; + sinc = (xap->nsamples << 16) / iSize; // calc freq by num / size + + if(xap->stereo) +{ + uint32_t * pS=(uint32_t *)xap->pcm; + uint32_t l=0; + + if(iXAPitch) + { + int32_t l1,l2;short s; + for(i=0;i<iSize;i++) + { + if(iUseInterpolation==2) + { + while(spos>=0x10000L) + { + l = *pS++; + gauss_window[gauss_ptr] = (short)LOWORD(l); + gauss_window[4+gauss_ptr] = (short)HIWORD(l); + gauss_ptr = (gauss_ptr+1) & 3; + spos -= 0x10000L; + } + vl = (spos >> 6) & ~3; + vr=(gauss[vl]*gvall0)&~2047; + vr+=(gauss[vl+1]*gvall(1))&~2047; + vr+=(gauss[vl+2]*gvall(2))&~2047; + vr+=(gauss[vl+3]*gvall(3))&~2047; + l= (vr >> 11) & 0xffff; + vr=(gauss[vl]*gvalr0)&~2047; + vr+=(gauss[vl+1]*gvalr(1))&~2047; + vr+=(gauss[vl+2]*gvalr(2))&~2047; + vr+=(gauss[vl+3]*gvalr(3))&~2047; + l |= vr << 5; + } + else + { + while(spos>=0x10000L) + { + l = *pS++; + spos -= 0x10000L; + } + } + + s=(short)LOWORD(l); + l1=s; + l1=(l1*iPlace)/iSize; + if(l1<-32767) l1=-32767; + if(l1> 32767) l1=32767; + s=(short)HIWORD(l); + l2=s; + l2=(l2*iPlace)/iSize; + if(l2<-32767) l2=-32767; + if(l2> 32767) l2=32767; + l=(l1&0xffff)|(l2<<16); + + *XAFeed++=l; + + if(XAFeed==XAEnd) XAFeed=XAStart; + if(XAFeed==XAPlay) + { + if(XAPlay!=XAStart) XAFeed=XAPlay-1; + break; + } + + spos += sinc; + } + } + else + { + for(i=0;i<iSize;i++) + { + if(iUseInterpolation==2) + { + while(spos>=0x10000L) + { + l = *pS++; + gauss_window[gauss_ptr] = (short)LOWORD(l); + gauss_window[4+gauss_ptr] = (short)HIWORD(l); + gauss_ptr = (gauss_ptr+1) & 3; + spos -= 0x10000L; + } + vl = (spos >> 6) & ~3; + vr=(gauss[vl]*gvall0)&~2047; + vr+=(gauss[vl+1]*gvall(1))&~2047; + vr+=(gauss[vl+2]*gvall(2))&~2047; + vr+=(gauss[vl+3]*gvall(3))&~2047; + l= (vr >> 11) & 0xffff; + vr=(gauss[vl]*gvalr0)&~2047; + vr+=(gauss[vl+1]*gvalr(1))&~2047; + vr+=(gauss[vl+2]*gvalr(2))&~2047; + vr+=(gauss[vl+3]*gvalr(3))&~2047; + l |= vr << 5; + } + else + { + while(spos>=0x10000L) + { + l = *pS++; + spos -= 0x10000L; + } + } + + *XAFeed++=l; + + if(XAFeed==XAEnd) XAFeed=XAStart; + if(XAFeed==XAPlay) + { + if(XAPlay!=XAStart) XAFeed=XAPlay-1; + break; + } + + spos += sinc; + } + } + } + else + { + unsigned short * pS=(unsigned short *)xap->pcm; + uint32_t l;short s=0; + + if(iXAPitch) + { + int32_t l1; + for(i=0;i<iSize;i++) + { + if(iUseInterpolation==2) + { + while(spos>=0x10000L) + { + gauss_window[gauss_ptr] = (short)*pS++; + gauss_ptr = (gauss_ptr+1) & 3; + spos -= 0x10000L; + } + vl = (spos >> 6) & ~3; + vr=(gauss[vl]*gvall0)&~2047; + vr+=(gauss[vl+1]*gvall(1))&~2047; + vr+=(gauss[vl+2]*gvall(2))&~2047; + vr+=(gauss[vl+3]*gvall(3))&~2047; + l1=s= vr >> 11; + l1 &= 0xffff; + } + else + { + while(spos>=0x10000L) + { + s = *pS++; + spos -= 0x10000L; + } + l1=s; + } + + l1=(l1*iPlace)/iSize; + if(l1<-32767) l1=-32767; + if(l1> 32767) l1=32767; + l=(l1&0xffff)|(l1<<16); + *XAFeed++=l; + + if(XAFeed==XAEnd) XAFeed=XAStart; + if(XAFeed==XAPlay) + { + if(XAPlay!=XAStart) XAFeed=XAPlay-1; + break; + } + + spos += sinc; + } + } + else + { + for(i=0;i<iSize;i++) + { + if(iUseInterpolation==2) + { + while(spos>=0x10000L) + { + gauss_window[gauss_ptr] = (short)*pS++; + gauss_ptr = (gauss_ptr+1) & 3; + spos -= 0x10000L; + } + vl = (spos >> 6) & ~3; + vr=(gauss[vl]*gvall0)&~2047; + vr+=(gauss[vl+1]*gvall(1))&~2047; + vr+=(gauss[vl+2]*gvall(2))&~2047; + vr+=(gauss[vl+3]*gvall(3))&~2047; + l=s= vr >> 11; + l &= 0xffff; + } + else + { + while(spos>=0x10000L) + { + s = *pS++; + spos -= 0x10000L; + } + l=s; + } + + *XAFeed++=(l|(l<<16)); + + if(XAFeed==XAEnd) XAFeed=XAStart; + if(XAFeed==XAPlay) + { + if(XAPlay!=XAStart) XAFeed=XAPlay-1; + break; + } + + spos += sinc; + } + } + } +} + +//////////////////////////////////////////////////////////////////////// +// FEED CDDA +//////////////////////////////////////////////////////////////////////// + +INLINE void FeedCDDA(unsigned char *pcm, int nBytes) +{ + while(nBytes>0) + { + if(CDDAFeed==CDDAEnd) CDDAFeed=CDDAStart; + while(CDDAFeed==CDDAPlay-1|| + (CDDAFeed==CDDAEnd-1&&CDDAPlay==CDDAStart)) + { + if (!iUseTimer) usleep(1000); + else return; + } + *CDDAFeed++=(*pcm | (*(pcm+1)<<8) | (*(pcm+2)<<16) | (*(pcm+3)<<24)); + nBytes-=4; + pcm+=4; + } +} + +#endif diff --git a/plugins/dfsound/xa.h b/plugins/dfsound/xa.h new file mode 100644 index 0000000..0928eba --- /dev/null +++ b/plugins/dfsound/xa.h @@ -0,0 +1,20 @@ +/***************************************************************************
+ xa.h - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+INLINE void MixXA(void);
+INLINE void FeedXA(xa_decode_t *xap);
+INLINE void FeedCDDA(unsigned char *pcm, int nBytes); |