diff options
Diffstat (limited to 'distrib/sdl-1.2.12/src/audio/dc')
| -rw-r--r-- | distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.c | 246 | ||||
| -rw-r--r-- | distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.h | 41 | ||||
| -rw-r--r-- | distrib/sdl-1.2.12/src/audio/dc/aica.c | 271 | ||||
| -rw-r--r-- | distrib/sdl-1.2.12/src/audio/dc/aica.h | 40 |
4 files changed, 0 insertions, 598 deletions
diff --git a/distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.c b/distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.c deleted file mode 100644 index a28ea5a..0000000 --- a/distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.c +++ /dev/null @@ -1,246 +0,0 @@ -/* - SDL - Simple DirectMedia Layer - Copyright (C) 1997-2006 Sam Lantinga - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public - License as published by the Free Software Foundation; either - version 2.1 of the License, or (at your option) any later version. - - This library 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 - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with this library; if not, write to the Free Software - Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - - Sam Lantinga - slouken@libsdl.org - -*/ -#include "SDL_config.h" - -/* Output dreamcast aica */ - -#include "SDL_timer.h" -#include "SDL_audio.h" -#include "../SDL_audiomem.h" -#include "../SDL_audio_c.h" -#include "../SDL_audiodev_c.h" -#include "SDL_dcaudio.h" - -#include "aica.h" -#include <dc/spu.h> - -/* Audio driver functions */ -static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec); -static void DCAUD_WaitAudio(_THIS); -static void DCAUD_PlayAudio(_THIS); -static Uint8 *DCAUD_GetAudioBuf(_THIS); -static void DCAUD_CloseAudio(_THIS); - -/* Audio driver bootstrap functions */ -static int DCAUD_Available(void) -{ - return 1; -} - -static void DCAUD_DeleteDevice(SDL_AudioDevice *device) -{ - SDL_free(device->hidden); - SDL_free(device); -} - -static SDL_AudioDevice *DCAUD_CreateDevice(int devindex) -{ - SDL_AudioDevice *this; - - /* Initialize all variables that we clean on shutdown */ - this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); - if ( this ) { - SDL_memset(this, 0, (sizeof *this)); - this->hidden = (struct SDL_PrivateAudioData *) - SDL_malloc((sizeof *this->hidden)); - } - if ( (this == NULL) || (this->hidden == NULL) ) { - SDL_OutOfMemory(); - if ( this ) { - SDL_free(this); - } - return(0); - } - SDL_memset(this->hidden, 0, (sizeof *this->hidden)); - - /* Set the function pointers */ - this->OpenAudio = DCAUD_OpenAudio; - this->WaitAudio = DCAUD_WaitAudio; - this->PlayAudio = DCAUD_PlayAudio; - this->GetAudioBuf = DCAUD_GetAudioBuf; - this->CloseAudio = DCAUD_CloseAudio; - - this->free = DCAUD_DeleteDevice; - - spu_init(); - - return this; -} - -AudioBootStrap DCAUD_bootstrap = { - "dcaudio", "Dreamcast AICA audio", - DCAUD_Available, DCAUD_CreateDevice -}; - -/* This function waits until it is possible to write a full sound buffer */ -static void DCAUD_WaitAudio(_THIS) -{ - if (this->hidden->playing) { - /* wait */ - while(aica_get_pos(0)/this->spec.samples == this->hidden->nextbuf) { - thd_pass(); - } - } -} - -#define SPU_RAM_BASE 0xa0800000 - -static void spu_memload_stereo8(int leftpos,int rightpos,void *src0,size_t size) -{ - uint8 *src = src0; - uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE); - uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE); - size = (size+7)/8; - while(size--) { - unsigned lval,rval; - lval = *src++; - rval = *src++; - lval|= (*src++)<<8; - rval|= (*src++)<<8; - lval|= (*src++)<<16; - rval|= (*src++)<<16; - lval|= (*src++)<<24; - rval|= (*src++)<<24; - g2_write_32(left++,lval); - g2_write_32(right++,rval); - g2_fifo_wait(); - } -} - -static void spu_memload_stereo16(int leftpos,int rightpos,void *src0,size_t size) -{ - uint16 *src = src0; - uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE); - uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE); - size = (size+7)/8; - while(size--) { - unsigned lval,rval; - lval = *src++; - rval = *src++; - lval|= (*src++)<<16; - rval|= (*src++)<<16; - g2_write_32(left++,lval); - g2_write_32(right++,rval); - g2_fifo_wait(); - } -} - -static void DCAUD_PlayAudio(_THIS) -{ - SDL_AudioSpec *spec = &this->spec; - unsigned int offset; - - if (this->hidden->playing) { - /* wait */ - while(aica_get_pos(0)/spec->samples == this->hidden->nextbuf) { - thd_pass(); - } - } - - offset = this->hidden->nextbuf*spec->size; - this->hidden->nextbuf^=1; - /* Write the audio data, checking for EAGAIN on broken audio drivers */ - if (spec->channels==1) { - spu_memload(this->hidden->leftpos+offset,this->hidden->mixbuf,this->hidden->mixlen); - } else { - offset/=2; - if ((this->spec.format&255)==8) { - spu_memload_stereo8(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen); - } else { - spu_memload_stereo16(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen); - } - } - - if (!this->hidden->playing) { - int mode; - this->hidden->playing = 1; - mode = (spec->format==AUDIO_S8)?SM_8BIT:SM_16BIT; - if (spec->channels==1) { - aica_play(0,mode,this->hidden->leftpos,0,spec->samples*2,spec->freq,255,128,1); - } else { - aica_play(0,mode,this->hidden->leftpos ,0,spec->samples*2,spec->freq,255,0,1); - aica_play(1,mode,this->hidden->rightpos,0,spec->samples*2,spec->freq,255,255,1); - } - } -} - -static Uint8 *DCAUD_GetAudioBuf(_THIS) -{ - return(this->hidden->mixbuf); -} - -static void DCAUD_CloseAudio(_THIS) -{ - aica_stop(0); - if (this->spec.channels==2) aica_stop(1); - if ( this->hidden->mixbuf != NULL ) { - SDL_FreeAudioMem(this->hidden->mixbuf); - this->hidden->mixbuf = NULL; - } -} - -static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec) -{ - Uint16 test_format = SDL_FirstAudioFormat(spec->format); - int valid_datatype = 0; - while ((!valid_datatype) && (test_format)) { - spec->format = test_format; - switch (test_format) { - /* only formats Dreamcast accepts... */ - case AUDIO_S8: - case AUDIO_S16LSB: - valid_datatype = 1; - break; - - default: - test_format = SDL_NextAudioFormat(); - break; - } - } - - if (!valid_datatype) { /* shouldn't happen, but just in case... */ - SDL_SetError("Unsupported audio format"); - return (-1); - } - - if (spec->channels > 2) - spec->channels = 2; /* no more than stereo on the Dreamcast. */ - - /* Update the fragment size as size in bytes */ - SDL_CalculateAudioSpec(spec); - - /* Allocate mixing buffer */ - this->hidden->mixlen = spec->size; - this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); - if ( this->hidden->mixbuf == NULL ) { - return(-1); - } - SDL_memset(this->hidden->mixbuf, spec->silence, spec->size); - this->hidden->leftpos = 0x11000; - this->hidden->rightpos = 0x11000+spec->size; - this->hidden->playing = 0; - this->hidden->nextbuf = 0; - - /* We're ready to rock and roll. :-) */ - return(0); -} diff --git a/distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.h b/distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.h deleted file mode 100644 index a5b01d3..0000000 --- a/distrib/sdl-1.2.12/src/audio/dc/SDL_dcaudio.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - SDL - Simple DirectMedia Layer - Copyright (C) 1997-2006 Sam Lantinga - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public - License as published by the Free Software Foundation; either - version 2.1 of the License, or (at your option) any later version. - - This library 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 - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with this library; if not, write to the Free Software - Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - - Sam Lantinga - slouken@libsdl.org -*/ -#include "SDL_config.h" - -#ifndef _SDL_dcaudio_h -#define _SDL_dcaudio_h - -#include "../SDL_sysaudio.h" - -/* Hidden "this" pointer for the video functions */ -#define _THIS SDL_AudioDevice *this - -struct SDL_PrivateAudioData { - /* The file descriptor for the audio device */ - Uint8 *mixbuf; - Uint32 mixlen; - int playing; - int leftpos,rightpos; - int nextbuf; -}; - -#endif /* _SDL_dcaudio_h */ diff --git a/distrib/sdl-1.2.12/src/audio/dc/aica.c b/distrib/sdl-1.2.12/src/audio/dc/aica.c deleted file mode 100644 index b6a1c93..0000000 --- a/distrib/sdl-1.2.12/src/audio/dc/aica.c +++ /dev/null @@ -1,271 +0,0 @@ -/* This file is part of the Dreamcast function library. - * Please see libdream.c for further details. - * - * (c)2000 Dan Potter - * modify BERO - */ -#include "aica.h" - -#include <arch/irq.h> -#include <dc/spu.h> - -/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */ -#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */ - -/* Some convienence macros */ -#define SNDREGADDR(x) (0xa0700000 + (x)) -#define CHNREGADDR(ch,x) SNDREGADDR(0x80*(ch)+(x)) - - -#define SNDREG32(x) (*(volatile unsigned long *)SNDREGADDR(x)) -#define SNDREG8(x) (*(volatile unsigned char *)SNDREGADDR(x)) -#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x)) -#define CHNREG8(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x)) - -#define G2_LOCK(OLD) \ - do { \ - if (!irq_inside_int()) \ - OLD = irq_disable(); \ - /* suspend any G2 DMA here... */ \ - while((*(volatile unsigned int *)0xa05f688c) & 0x20) \ - ; \ - } while(0) - -#define G2_UNLOCK(OLD) \ - do { \ - /* resume any G2 DMA here... */ \ - if (!irq_inside_int()) \ - irq_restore(OLD); \ - } while(0) - - -void aica_init() { - int i, j, old = 0; - - /* Initialize AICA channels */ - G2_LOCK(old); - SNDREG32(0x2800) = 0x0000; - - for (i=0; i<64; i++) { - for (j=0; j<0x80; j+=4) { - if ((j&31)==0) g2_fifo_wait(); - CHNREG32(i, j) = 0; - } - g2_fifo_wait(); - CHNREG32(i,0) = 0x8000; - CHNREG32(i,20) = 0x1f; - } - - SNDREG32(0x2800) = 0x000f; - g2_fifo_wait(); - G2_UNLOCK(old); -} - -/* Translates a volume from linear form to logarithmic form (required by - the AICA chip */ -/* int logs[] = { - -0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103, -105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127, -129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145, -146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159, -160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171, -172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182, -182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191, -191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199, -200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207, -208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215, -215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222, -222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228, -228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234, -234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240, -240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245, -246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251, -251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255 - -}; */ - -const static unsigned char logs[] = { - 0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61, - 63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88, - 90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106, - 108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121, - 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134, - 135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146, - 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156, - 157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167, - 167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176, - 177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185, - 186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194, - 195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202, - 203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210, - 211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218, - 219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225, - 226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233, - 233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240, - 240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246, - 247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255 -}; - -/* For the moment this is going to have to suffice, until we really - figure out what these mean. */ -#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f))) -#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)]) -//#define AICA_VOL(x) (0xff - logs[x&255]) - -static inline unsigned AICA_FREQ(unsigned freq) { - unsigned long freq_lo, freq_base = 5644800; - int freq_hi = 7; - - /* Need to convert frequency to floating point format - (freq_hi is exponent, freq_lo is mantissa) - Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */ - while (freq < freq_base && freq_hi > -8) { - freq_base >>= 1; - --freq_hi; - } - while (freq < freq_base && freq_hi > -8) { - freq_base >>= 1; - freq_hi--; - } - freq_lo = (freq<<10) / freq_base; - return (freq_hi << 11) | (freq_lo & 1023); -} - -/* Sets up a sound channel completely. This is generally good if you want - a quick and dirty way to play notes. If you want a more comprehensive - set of routines (more like PC wavetable cards) see below. - - ch is the channel to play on (0 - 63) - smpptr is the pointer to the sound data; if you're running off the - SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just - ptr. Basically, it's an offset into sound ram. - mode is one of the mode constants (16 bit, 8 bit, ADPCM) - nsamp is the number of samples to play (not number of bytes!) - freq is the sampling rate of the sound - vol is the volume, 0 to 0xff (0xff is louder) - pan is a panning constant -- 0 is left, 128 is center, 255 is right. - - This routine (and the similar ones) owe a lot to Marcus' sound example -- - I hadn't gotten quite this far into dissecting the individual regs yet. */ -void aica_play(int ch,int mode,unsigned long smpptr,int loopst,int loopend,int freq,int vol,int pan,int loopflag) { -/* int i; -*/ - int val; - int old = 0; - - /* Stop the channel (if it's already playing) */ - aica_stop(ch); - /* doesn't seem to be needed, but it's here just in case */ -/* - for (i=0; i<256; i++) { - asm("nop"); - asm("nop"); - asm("nop"); - asm("nop"); - } -*/ - G2_LOCK(old); - /* Envelope setup. The first of these is the loop point, - e.g., where the sample starts over when it loops. The second - is the loop end. This is the full length of the sample when - you are not looping, or the loop end point when you are (though - storing more than that is a waste of memory if you're not doing - volume enveloping). */ - CHNREG32(ch, 8) = loopst & 0xffff; - CHNREG32(ch, 12) = loopend & 0xffff; - - /* Write resulting values */ - CHNREG32(ch, 24) = AICA_FREQ(freq); - - /* Set volume, pan, and some other things that we don't know what - they do =) */ - CHNREG32(ch, 36) = AICA_PAN(pan) | (0xf<<8); - /* Convert the incoming volume and pan into hardware values */ - /* Vol starts at zero so we can ramp */ - vol = AICA_VOL(vol); - CHNREG32(ch, 40) = 0x24 | (vol<<8); - /* Convert the incoming volume and pan into hardware values */ - /* Vol starts at zero so we can ramp */ - - /* If we supported volume envelopes (which we don't yet) then - this value would set that up. The top 4 bits determine the - envelope speed. f is the fastest, 1 is the slowest, and 0 - seems to be an invalid value and does weird things). The - default (below) sets it into normal mode (play and terminate/loop). - CHNREG32(ch, 16) = 0xf010; - */ - CHNREG32(ch, 16) = 0x1f; /* No volume envelope */ - - - /* Set sample format, buffer address, and looping control. If - 0x0200 mask is set on reg 0, the sample loops infinitely. If - it's not set, the sample plays once and terminates. We'll - also set the bits to start playback here. */ - CHNREG32(ch, 4) = smpptr & 0xffff; - val = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16); - if (loopflag) val|=0x200; - - CHNREG32(ch, 0) = val; - - G2_UNLOCK(old); - - /* Enable playback */ - /* CHNREG32(ch, 0) |= 0xc000; */ - g2_fifo_wait(); - -#if 0 - for (i=0xff; i>=vol; i--) { - if ((i&7)==0) g2_fifo_wait(); - CHNREG32(ch, 40) = 0x24 | (i<<8);; - } - - g2_fifo_wait(); -#endif -} - -/* Stop the sound on a given channel */ -void aica_stop(int ch) { - g2_write_32(CHNREGADDR(ch, 0),(g2_read_32(CHNREGADDR(ch, 0)) & ~0x4000) | 0x8000); - g2_fifo_wait(); -} - - -/* The rest of these routines can change the channel in mid-stride so you - can do things like vibrato and panning effects. */ - -/* Set channel volume */ -void aica_vol(int ch,int vol) { -// g2_write_8(CHNREGADDR(ch, 41),AICA_VOL(vol)); - g2_write_32(CHNREGADDR(ch, 40),(g2_read_32(CHNREGADDR(ch, 40))&0xffff00ff)|(AICA_VOL(vol)<<8) ); - g2_fifo_wait(); -} - -/* Set channel pan */ -void aica_pan(int ch,int pan) { -// g2_write_8(CHNREGADDR(ch, 36),AICA_PAN(pan)); - g2_write_32(CHNREGADDR(ch, 36),(g2_read_32(CHNREGADDR(ch, 36))&0xffffff00)|(AICA_PAN(pan)) ); - g2_fifo_wait(); -} - -/* Set channel frequency */ -void aica_freq(int ch,int freq) { - g2_write_32(CHNREGADDR(ch, 24),AICA_FREQ(freq)); - g2_fifo_wait(); -} - -/* Get channel position */ -int aica_get_pos(int ch) { -#if 1 - /* Observe channel ch */ - g2_write_32(SNDREGADDR(0x280c),(g2_read_32(SNDREGADDR(0x280c))&0xffff00ff) | (ch<<8)); - g2_fifo_wait(); - /* Update position counters */ - return g2_read_32(SNDREGADDR(0x2814)) & 0xffff; -#else - /* Observe channel ch */ - g2_write_8(SNDREGADDR(0x280d),ch); - /* Update position counters */ - return g2_read_32(SNDREGADDR(0x2814)) & 0xffff; -#endif -} diff --git a/distrib/sdl-1.2.12/src/audio/dc/aica.h b/distrib/sdl-1.2.12/src/audio/dc/aica.h deleted file mode 100644 index 93155d2..0000000 --- a/distrib/sdl-1.2.12/src/audio/dc/aica.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - SDL - Simple DirectMedia Layer - Copyright (C) 1997-2006 Sam Lantinga - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public - License as published by the Free Software Foundation; either - version 2.1 of the License, or (at your option) any later version. - - This library 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 - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with this library; if not, write to the Free Software - Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - - Sam Lantinga - slouken@libsdl.org -*/ -#include "SDL_config.h" - -#ifndef _AICA_H_ -#define _AICA_H_ - -#define AICA_MEM 0xa0800000 - -#define SM_8BIT 1 -#define SM_16BIT 0 -#define SM_ADPCM 2 - -void aica_play(int ch,int mode,unsigned long smpptr,int looptst,int loopend,int freq,int vol,int pan,int loopflag); -void aica_stop(int ch); -void aica_vol(int ch,int vol); -void aica_pan(int ch,int pan); -void aica_freq(int ch,int freq); -int aica_get_pos(int ch); - -#endif |