diff options
| -rw-r--r-- | Makefile.target | 39 | ||||
| -rw-r--r-- | audio/audio.c | 935 | ||||
| -rw-r--r-- | audio/audio.h | 188 | ||||
| -rw-r--r-- | audio/fmodaudio.c | 457 | ||||
| -rw-r--r-- | audio/fmodaudio.h | 39 | ||||
| -rw-r--r-- | audio/mixeng.c | 255 | ||||
| -rw-r--r-- | audio/mixeng.h | 39 | ||||
| -rw-r--r-- | audio/mixeng_template.h | 111 | ||||
| -rw-r--r-- | audio/ossaudio.c | 466 | ||||
| -rw-r--r-- | audio/ossaudio.h | 40 | ||||
| -rw-r--r-- | audio/sdlaudio.c | 323 | ||||
| -rw-r--r-- | audio/sdlaudio.h | 34 | ||||
| -rw-r--r-- | audio/wavaudio.c | 200 | ||||
| -rw-r--r-- | audio/wavaudio.h | 38 | ||||
| -rw-r--r-- | hw/adlib.c | 309 | ||||
| -rw-r--r-- | hw/dma.c | 215 | ||||
| -rw-r--r-- | hw/fdc.c | 31 | ||||
| -rw-r--r-- | hw/fmopl.c | 1390 | ||||
| -rw-r--r-- | hw/fmopl.h | 174 | ||||
| -rw-r--r-- | hw/pc.c | 3 | ||||
| -rw-r--r-- | hw/sb16.c | 1412 | ||||
| -rw-r--r-- | oss.c | 978 | ||||
| -rw-r--r-- | vl.c | 6 | ||||
| -rw-r--r-- | vl.h | 30 |
24 files changed, 6038 insertions, 1674 deletions
diff --git a/Makefile.target b/Makefile.target index 6ac8d9f1b0..280ffa1b3c 100644 --- a/Makefile.target +++ b/Makefile.target @@ -1,7 +1,17 @@ include config.mak +#After enabling Adlib and/or FMOD rebuild QEMU from scratch +#Uncomment following for adlib support +#USE_ADLIB=1 + +#Uncomment following and specify proper paths/names for FMOD support +#USE_FMOD=1 +#FMOD_INCLUDE=/net/include/fmod +#FMOD_LIBPATH=/net/lib +#FMOD_VERSION=3.74 + TARGET_PATH=$(SRC_PATH)/target-$(TARGET_ARCH) -VPATH=$(SRC_PATH):$(TARGET_PATH):$(SRC_PATH)/hw +VPATH=$(SRC_PATH):$(TARGET_PATH):$(SRC_PATH)/hw:$(SRC_PATH)/audio DEFINES=-I. -I$(TARGET_PATH) -I$(SRC_PATH) ifdef CONFIG_USER_ONLY VPATH+=:$(SRC_PATH)/linux-user @@ -267,16 +277,31 @@ endif VL_OBJS=vl.o osdep.o block.o readline.o monitor.o pci.o console.o VL_OBJS+=block-cow.o block-qcow.o aes.o block-vmdk.o block-cloop.o +SOUND_HW = sb16.o +AUDIODRV = audio.o ossaudio.o sdlaudio.o wavaudio.o + +ifeq ($(USE_ADLIB),1) +SOUND_HW += fmopl.o adlib.o +audio.o: DEFINES := -DUSE_ADLIB $(DEFINES) +endif + +ifeq ($(USE_FMOD),1) +AUDIODRV += fmodaudio.o +audio.o fmodaudio.o: DEFINES := -DUSE_FMOD_AUDIO -I$(FMOD_INCLUDE) $(DEFINES) +LDFLAGS += -L$(FMOD_LIBPATH) -Wl,-rpath,$(FMOD_LIBPATH) +LIBS += -lfmod-$(FMOD_VERSION) +endif + ifeq ($(TARGET_ARCH), i386) # Hardware support -VL_OBJS+= ide.o ne2000.o pckbd.o vga.o sb16.o dma.o oss.o -VL_OBJS+= fdc.o mc146818rtc.o serial.o i8259.o i8254.o pc.o -VL_OBJS+= cirrus_vga.o +VL_OBJS+= ide.o ne2000.o pckbd.o vga.o $(SOUND_HW) dma.o $(AUDIODRV) +VL_OBJS+= fdc.o mc146818rtc.o serial.o i8259.o i8254.o pc.o +VL_OBJS+= cirrus_vga.o mixeng.o endif ifeq ($(TARGET_ARCH), ppc) -VL_OBJS+= ppc.o ide.o ne2000.o pckbd.o vga.o sb16.o dma.o oss.o +VL_OBJS+= ppc.o ide.o ne2000.o pckbd.o vga.o $(SOUND_HW) dma.o $(AUDIODRV) VL_OBJS+= mc146818rtc.o serial.o i8259.o i8254.o fdc.o m48t59.o -VL_OBJS+= ppc_prep.o ppc_chrp.o cuda.o adb.o openpic.o +VL_OBJS+= ppc_prep.o ppc_chrp.o cuda.o adb.o openpic.o mixeng.o endif ifeq ($(TARGET_ARCH), sparc) VL_OBJS+= sun4m.o tcx.o lance.o iommu.o sched.o m48t08.o magic-load.o timer.o @@ -360,6 +385,8 @@ op.o: op.c op_template.h op_mem.h op_helper.o: op_helper_mem.h endif +mixeng.o: mixeng.c mixeng.h mixeng_template.h + %.o: %.c $(CC) $(CFLAGS) $(DEFINES) -c -o $@ $< diff --git a/audio/audio.c b/audio/audio.c new file mode 100644 index 0000000000..f55e1a28cc --- /dev/null +++ b/audio/audio.c @@ -0,0 +1,935 @@ +/* + * QEMU Audio subsystem + * + * Copyright (c) 2003-2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include <assert.h> +#include <limits.h> +#include "vl.h" + +#define AUDIO_CAP "audio" +#include "audio/audio.h" + +#define USE_SDL_AUDIO +#define USE_WAV_AUDIO + +#if defined __linux__ || (defined _BSD && !defined __APPLE__) +#define USE_OSS_AUDIO +#endif + +#ifdef USE_OSS_AUDIO +#include "audio/ossaudio.h" +#endif + +#ifdef USE_SDL_AUDIO +#include "audio/sdlaudio.h" +#endif + +#ifdef USE_WAV_AUDIO +#include "audio/wavaudio.h" +#endif + +#ifdef USE_FMOD_AUDIO +#include "audio/fmodaudio.h" +#endif + +#define QC_AUDIO_DRV "QEMU_AUDIO_DRV" +#define QC_VOICES "QEMU_VOICES" +#define QC_FIXED_FORMAT "QEMU_FIXED_FORMAT" +#define QC_FIXED_FREQ "QEMU_FIXED_FREQ" + +extern void SB16_init (void); + +#ifdef USE_ADLIB +extern void Adlib_init (void); +#endif + +#ifdef USE_GUS +extern void GUS_init (void); +#endif + +static void (*hw_ctors[]) (void) = { + SB16_init, +#ifdef USE_ADLIB + Adlib_init, +#endif +#ifdef USE_GUS + GUS_init, +#endif + NULL +}; + +static HWVoice *hw_voice; + +AudioState audio_state = { + 1, /* use fixed settings */ + 44100, /* fixed frequency */ + 2, /* fixed channels */ + AUD_FMT_S16, /* fixed format */ + 1, /* number of hw voices */ + -1 /* voice size */ +}; + +/* http://www.df.lth.se/~john_e/gems/gem002d.html */ +/* http://www.multi-platforms.com/Tips/PopCount.htm */ +uint32_t popcount (uint32_t u) +{ + u = ((u&0x55555555) + ((u>>1)&0x55555555)); + u = ((u&0x33333333) + ((u>>2)&0x33333333)); + u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f)); + u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff)); + u = ( u&0x0000ffff) + (u>>16); + return u; +} + +inline uint32_t lsbindex (uint32_t u) +{ + return popcount ((u&-u)-1); +} + +int audio_get_conf_int (const char *key, int defval) +{ + int val = defval; + char *strval; + + strval = getenv (key); + if (strval) { + val = atoi (strval); + } + + return val; +} + +const char *audio_get_conf_str (const char *key, const char *defval) +{ + const char *val = getenv (key); + if (!val) + return defval; + else + return val; +} + +void audio_log (const char *fmt, ...) +{ + va_list ap; + va_start (ap, fmt); + vfprintf (stderr, fmt, ap); + va_end (ap); +} + +/* + * Soft Voice + */ +void pcm_sw_free_resources (SWVoice *sw) +{ + if (sw->buf) qemu_free (sw->buf); + if (sw->rate) st_rate_stop (sw->rate); + sw->buf = NULL; + sw->rate = NULL; +} + +int pcm_sw_alloc_resources (SWVoice *sw) +{ + sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t)); + if (!sw->buf) + return -1; + + sw->rate = st_rate_start (sw->freq, sw->hw->freq); + if (!sw->rate) { + qemu_free (sw->buf); + sw->buf = NULL; + return -1; + } + return 0; +} + +void pcm_sw_fini (SWVoice *sw) +{ + pcm_sw_free_resources (sw); +} + +int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq, + int nchannels, audfmt_e fmt) +{ + int bits = 8, sign = 0; + + switch (fmt) { + case AUD_FMT_S8: + sign = 1; + case AUD_FMT_U8: + break; + + case AUD_FMT_S16: + sign = 1; + case AUD_FMT_U16: + bits = 16; + break; + } + + sw->hw = hw; + sw->freq = freq; + sw->fmt = fmt; + sw->nchannels = nchannels; + sw->shift = (nchannels == 2) + (bits == 16); + sw->align = (1 << sw->shift) - 1; + sw->left = 0; + sw->pos = 0; + sw->wpos = 0; + sw->live = 0; + sw->ratio = (sw->hw->freq * ((int64_t) INT_MAX)) / sw->freq; + sw->bytes_per_second = sw->freq << sw->shift; + sw->conv = mixeng_conv[nchannels == 2][sign][bits == 16]; + + pcm_sw_free_resources (sw); + return pcm_sw_alloc_resources (sw); +} + +/* Hard voice */ +void pcm_hw_free_resources (HWVoice *hw) +{ + if (hw->mix_buf) + qemu_free (hw->mix_buf); + hw->mix_buf = NULL; +} + +int pcm_hw_alloc_resources (HWVoice *hw) +{ + hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t)); + if (!hw->mix_buf) + return -1; + return 0; +} + + +void pcm_hw_fini (HWVoice *hw) +{ + if (hw->active) { + ldebug ("pcm_hw_fini: %d %d %d\n", hw->freq, hw->nchannels, hw->fmt); + pcm_hw_free_resources (hw); + hw->pcm_ops->fini (hw); + memset (hw, 0, audio_state.drv->voice_size); + } +} + +void pcm_hw_gc (HWVoice *hw) +{ + if (hw->nb_voices) + return; + + pcm_hw_fini (hw); +} + +int pcm_hw_get_live (HWVoice *hw) +{ + int i, alive = 0, live = hw->samples; + + for (i = 0; i < hw->nb_voices; i++) { + if (hw->pvoice[i]->live) { + live = audio_MIN (hw->pvoice[i]->live, live); + alive += 1; + } + } + + if (alive) + return live; + else + return -1; +} + +int pcm_hw_get_live2 (HWVoice *hw, int *nb_active) +{ + int i, alive = 0, live = hw->samples; + + *nb_active = 0; + for (i = 0; i < hw->nb_voices; i++) { + if (hw->pvoice[i]->live) { + if (hw->pvoice[i]->live < live) { + *nb_active = hw->pvoice[i]->active != 0; + live = hw->pvoice[i]->live; + } + alive += 1; + } + } + + if (alive) + return live; + else + return -1; +} + +void pcm_hw_dec_live (HWVoice *hw, int decr) +{ + int i; + + for (i = 0; i < hw->nb_voices; i++) { + if (hw->pvoice[i]->live) { + hw->pvoice[i]->live -= decr; + } + } +} + +void pcm_hw_clear (HWVoice *hw, void *buf, int len) +{ + if (!len) + return; + + switch (hw->fmt) { + case AUD_FMT_S16: + case AUD_FMT_S8: + memset (buf, len << hw->shift, 0x00); + break; + + case AUD_FMT_U8: + memset (buf, len << hw->shift, 0x80); + break; + + case AUD_FMT_U16: + { + unsigned int i; + uint16_t *p = buf; + int shift = hw->nchannels - 1; + + for (i = 0; i < len << shift; i++) { + p[i] = INT16_MAX; + } + } + break; + } +} + +int pcm_hw_write (SWVoice *sw, void *buf, int size) +{ + int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck; + int ret = 0, pos = 0; + if (!sw) + return size; + + hwsamples = sw->hw->samples; + samples = size >> sw->shift; + + if (!sw->live) { + sw->wpos = sw->hw->rpos; + } + wpos = sw->wpos; + live = sw->live; + dead = hwsamples - live; + swlim = (dead * ((int64_t) INT_MAX)) / sw->ratio; + swlim = audio_MIN (swlim, samples); + + ldebug ("size=%d live=%d dead=%d swlim=%d wpos=%d\n", + size, live, dead, swlim, wpos); + if (swlim) + sw->conv (sw->buf, buf, swlim); + + while (swlim) { + dead = hwsamples - live; + left = hwsamples - wpos; + blck = audio_MIN (dead, left); + if (!blck) { + /* dolog ("swlim=%d\n", swlim); */ + break; + } + isamp = swlim; + osamp = blck; + st_rate_flow (sw->rate, sw->buf + pos, sw->hw->mix_buf + wpos, &isamp, &osamp); + ret += isamp; + swlim -= isamp; + pos += isamp; + live += osamp; + wpos = (wpos + osamp) % hwsamples; + } + + sw->wpos = wpos; + sw->live = live; + return ret << sw->shift; +} + +int pcm_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt) +{ + int sign = 0, bits = 8; + + pcm_hw_fini (hw); + ldebug ("pcm_hw_init: %d %d %d\n", freq, nchannels, fmt); + if (hw->pcm_ops->init (hw, freq, nchannels, fmt)) { + memset (hw, 0, audio_state.drv->voice_size); + return -1; + } + + switch (hw->fmt) { + case AUD_FMT_S8: + sign = 1; + case AUD_FMT_U8: + break; + + case AUD_FMT_S16: + sign = 1; + case AUD_FMT_U16: + bits = 16; + break; + } + + hw->nb_voices = 0; + hw->active = 1; + hw->shift = (hw->nchannels == 2) + (bits == 16); + hw->bytes_per_second = hw->freq << hw->shift; + hw->align = (1 << hw->shift) - 1; + hw->samples = hw->bufsize >> hw->shift; + hw->clip = mixeng_clip[hw->nchannels == 2][sign][bits == 16]; + if (pcm_hw_alloc_resources (hw)) { + pcm_hw_fini (hw); + return -1; + } + return 0; +} + +static int dist (void *hw) +{ + if (hw) { + return (((uint8_t *) hw - (uint8_t *) hw_voice) + / audio_state.voice_size) + 1; + } + else { + return 0; + } +} + +#define ADVANCE(hw) hw ? advance (hw, audio_state.voice_size) : hw_voice + +HWVoice *pcm_hw_find_any (HWVoice *hw) +{ + int i = dist (hw); + for (; i < audio_state.nb_hw_voices; i++) { + hw = ADVANCE (hw); + return hw; + } + return NULL; +} + +HWVoice *pcm_hw_find_any_active (HWVoice *hw) +{ + int i = dist (hw); + for (; i < audio_state.nb_hw_voices; i++) { + hw = ADVANCE (hw); + if (hw->active) + return hw; + } + return NULL; +} + +HWVoice *pcm_hw_find_any_active_enabled (HWVoice *hw) +{ + int i = dist (hw); + for (; i < audio_state.nb_hw_voices; i++) { + hw = ADVANCE (hw); + if (hw->active && hw->enabled) + return hw; + } + return NULL; +} + +HWVoice *pcm_hw_find_any_passive (HWVoice *hw) +{ + int i = dist (hw); + for (; i < audio_state.nb_hw_voices; i++) { + hw = ADVANCE (hw); + if (!hw->active) + return hw; + } + return NULL; +} + +HWVoice *pcm_hw_find_specific (HWVoice *hw, int freq, + int nchannels, audfmt_e fmt) +{ + while ((hw = pcm_hw_find_any_active (hw))) { + if (hw->freq == freq && + hw->nchannels == nchannels && + hw->fmt == fmt) + return hw; + } + return NULL; +} + +HWVoice *pcm_hw_add (int freq, int nchannels, audfmt_e fmt) +{ + HWVoice *hw; + + if (audio_state.fixed_format) { + freq = audio_state.fixed_freq; + nchannels = audio_state.fixed_channels; + fmt = audio_state.fixed_fmt; + } + + hw = pcm_hw_find_specific (NULL, freq, nchannels, fmt); + + if (hw) + return hw; + + hw = pcm_hw_find_any_passive (NULL); + if (hw) { + hw->pcm_ops = audio_state.drv->pcm_ops; + if (!hw->pcm_ops) + return NULL; + + if (pcm_hw_init (hw, freq, nchannels, fmt)) { + pcm_hw_gc (hw); + return NULL; + } + else + return hw; + } + + return pcm_hw_find_any (NULL); +} + +int pcm_hw_add_sw (HWVoice *hw, SWVoice *sw) +{ + SWVoice **pvoice = qemu_mallocz ((hw->nb_voices + 1) * sizeof (sw)); + if (!pvoice) + return -1; + + memcpy (pvoice, hw->pvoice, hw->nb_voices * sizeof (sw)); + qemu_free (hw->pvoice); + hw->pvoice = pvoice; + hw->pvoice[hw->nb_voices++] = sw; + return 0; +} + +int pcm_hw_del_sw (HWVoice *hw, SWVoice *sw) +{ + int i, j; + if (hw->nb_voices > 1) { + SWVoice **pvoice = qemu_mallocz ((hw->nb_voices - 1) * sizeof (sw)); + + if (!pvoice) { + dolog ("Can not maintain consistent state (not enough memory)\n"); + return -1; + } + + for (i = 0, j = 0; i < hw->nb_voices; i++) { + if (j >= hw->nb_voices - 1) { + dolog ("Can not maintain consistent state " + "(invariant violated)\n"); + return -1; + } + if (hw->pvoice[i] != sw) + pvoice[j++] = hw->pvoice[i]; + } + qemu_free (hw->pvoice); + hw->pvoice = pvoice; + hw->nb_voices -= 1; + } + else { + qemu_free (hw->pvoice); + hw->pvoice = NULL; + hw->nb_voices = 0; + } + return 0; +} + +SWVoice *pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt) +{ + SWVoice *sw; + HWVoice *hw; + + sw = qemu_mallocz (sizeof (*sw)); + if (!sw) + goto err1; + + hw = pcm_hw_add (freq, nchannels, fmt); + if (!hw) + goto err2; + + if (pcm_hw_add_sw (hw, sw)) + goto err3; + + if (pcm_sw_init (sw, hw, freq, nchannels, fmt)) + goto err4; + + return sw; + +err4: + pcm_hw_del_sw (hw, sw); +err3: + pcm_hw_gc (hw); +err2: + qemu_free (sw); +err1: + return NULL; +} + +SWVoice *AUD_open (SWVoice *sw, const char *name, + int freq, int nchannels, audfmt_e fmt) +{ + if (!audio_state.drv) { + return NULL; + } + + if (sw && freq == sw->freq && sw->nchannels == nchannels && sw->fmt == fmt) { + return sw; + } + + if (sw) { + ldebug ("Different format %s %d %d %d\n", + name, + sw->freq == freq, + sw->nchannels == nchannels, + sw->fmt == fmt); + } + + if (nchannels != 1 && nchannels != 2) { + dolog ("Bogus channel count %d for voice %s\n", nchannels, name); + return NULL; + } + + if (!audio_state.fixed_format && sw) { + pcm_sw_fini (sw); + pcm_hw_del_sw (sw->hw, sw); + pcm_hw_gc (sw->hw); + if (sw->name) { + qemu_free (sw->name); + sw->name = NULL; + } + qemu_free (sw); + sw = NULL; + } + + if (sw) { + HWVoice *hw = sw->hw; + if (!hw) { + dolog ("Internal logic error voice %s has no hardware store\n", + name); + return sw; + } + + if (pcm_sw_init (sw, hw, freq, nchannels, fmt)) { + pcm_sw_fini (sw); + pcm_hw_del_sw (hw, sw); + pcm_hw_gc (hw); + if (sw->name) { + qemu_free (sw->name); + sw->name = NULL; + } + qemu_free (sw); + return NULL; + } + } + else { + sw = pcm_create_voice_pair (freq, nchannels, fmt); + if (!sw) { + dolog ("Failed to create voice %s\n", name); + return NULL; + } + } + + if (sw->name) { + qemu_free (sw->name); + sw->name = NULL; + } + sw->name = qemu_strdup (name); + return sw; +} + +int AUD_write (SWVoice *sw, void *buf, int size) +{ + int bytes; + + if (!sw->hw->enabled) + dolog ("Writing to disabled voice %s\n", sw->name); + bytes = sw->hw->pcm_ops->write (sw, buf, size); + return bytes; +} + +void AUD_run (void) +{ + HWVoice *hw = NULL; + + while ((hw = pcm_hw_find_any_active_enabled (hw))) { + int i; + if (hw->pending_disable && pcm_hw_get_live (hw) <= 0) { + hw->enabled = 0; + hw->pcm_ops->ctl (hw, VOICE_DISABLE); + for (i = 0; i < hw->nb_voices; i++) { + hw->pvoice[i]->live = 0; + /* hw->pvoice[i]->old_ticks = 0; */ + } + continue; + } + + hw->pcm_ops->run (hw); + assert (hw->rpos < hw->samples); + for (i = 0; i < hw->nb_voices; i++) { + SWVoice *sw = hw->pvoice[i]; + if (!sw->active && !sw->live && sw->old_ticks) { + int64_t delta = qemu_get_clock (vm_clock) - sw->old_ticks; + if (delta > audio_state.ticks_threshold) { + ldebug ("resetting old_ticks for %s\n", sw->name); + sw->old_ticks = 0; + } + } + } + } +} + +int AUD_get_free (SWVoice *sw) +{ + int free; + + if (!sw) + return 4096; + + free = ((sw->hw->samples - sw->live) << sw->hw->shift) * sw->ratio + / INT_MAX; + + free &= ~sw->hw->align; + if (!free) return 0; + + return free; +} + +int AUD_get_buffer_size (SWVoice *sw) +{ + return sw->hw->bufsize; +} + +void AUD_adjust (SWVoice *sw, int bytes) +{ + if (!sw) + return; + sw->old_ticks += (ticks_per_sec * (int64_t) bytes) / sw->bytes_per_second; +} + +void AUD_reset (SWVoice *sw) +{ + sw->active = 0; + sw->old_ticks = 0; +} + +int AUD_calc_elapsed (SWVoice *sw) +{ + int64_t now, delta, bytes; + int dead, swlim; + + if (!sw) + return 0; + + now = qemu_get_clock (vm_clock); + delta = now - sw->old_ticks; + bytes = (delta * sw->bytes_per_second) / ticks_per_sec; + if (delta < 0) { + dolog ("whoops delta(<0)=%lld\n", delta); + return 0; + } + + dead = sw->hw->samples - sw->live; + swlim = ((dead * (int64_t) INT_MAX) / sw->ratio); + + if (bytes > swlim) { + return swlim; + } + else { + return bytes; + } +} + +void AUD_enable (SWVoice *sw, int on) +{ + int i; + HWVoice *hw; + + if (!sw) + return; + + hw = sw->hw; + if (on) { + if (!sw->live) + sw->wpos = sw->hw->rpos; + if (!sw->old_ticks) { + sw->old_ticks = qemu_get_clock (vm_clock); + } + } + + if (sw->active != on) { + if (on) { + hw->pending_disable = 0; + if (!hw->enabled) { + hw->enabled = 1; + for (i = 0; i < hw->nb_voices; i++) { + ldebug ("resetting voice\n"); + sw = hw->pvoice[i]; + sw->old_ticks = qemu_get_clock (vm_clock); + } + hw->pcm_ops->ctl (hw, VOICE_ENABLE); + } + } + else { + if (hw->enabled && !hw->pending_disable) { + int nb_active = 0; + for (i = 0; i < hw->nb_voices; i++) { + nb_active += hw->pvoice[i]->active != 0; + } + + if (nb_active == 1) { + hw->pending_disable = 1; + } + } + } + sw->active = on; + } +} + +static struct audio_output_driver *drvtab[] = { +#ifdef USE_OSS_AUDIO + &oss_output_driver, +#endif +#ifdef USE_FMOD_AUDIO + &fmod_output_driver, +#endif +#ifdef USE_SDL_AUDIO + &sdl_output_driver, +#endif +#ifdef USE_WAV_AUDIO + &wav_output_driver, +#endif +}; + +static int voice_init (struct audio_output_driver *drv) +{ + audio_state.opaque = drv->init (); + if (audio_state.opaque) { + if (audio_state.nb_hw_voices > drv->max_voices) { + dolog ("`%s' does not support %d multiple hardware channels\n" + "Resetting to %d\n", + drv->name, audio_state.nb_hw_voices, drv->max_voices); + audio_state.nb_hw_voices = drv->max_voices; + } + hw_voice = qemu_mallocz (audio_state.nb_hw_voices * drv->voice_size); + if (hw_voice) { + audio_state.drv = drv; + return 1; + } + else { + dolog ("Not enough memory for %d `%s' voices (each %d bytes)\n", + audio_state.nb_hw_voices, drv->name, drv->voice_size); + drv->fini (audio_state.opaque); + return 0; + } + } + else { + dolog ("Could not init `%s' audio\n", drv->name); + return 0; + } +} + +static void audio_vm_stop_handler (void *opaque, int reason) +{ + HWVoice *hw = NULL; + + while ((hw = pcm_hw_find_any (hw))) { + if (!hw->pcm_ops) + continue; + + hw->pcm_ops->ctl (hw, reason ? VOICE_ENABLE : VOICE_DISABLE); + } +} + +static void audio_atexit (void) +{ + HWVoice *hw = NULL; + + while ((hw = pcm_hw_find_any (hw))) { + if (!hw->pcm_ops) + continue; + + hw->pcm_ops->ctl (hw, VOICE_DISABLE); + hw->pcm_ops->fini (hw); + } + audio_state.drv->fini (audio_state.opaque); +} + +static void audio_save (QEMUFile *f, void *opaque) +{ +} + +static int audio_load (QEMUFile *f, void *opaque, int version_id) +{ + if (version_id != 1) + return -EINVAL; + + return 0; +} + +void AUD_init (void) +{ + int i; + int done = 0; + const char *drvname; + + audio_state.fixed_format = + !!audio_get_conf_int (QC_FIXED_FORMAT, audio_state.fixed_format); + audio_state.fixed_freq = + audio_get_conf_int (QC_FIXED_FREQ, audio_state.fixed_freq); + audio_state.nb_hw_voices = + audio_get_conf_int (QC_VOICES, audio_state.nb_hw_voices); + + if (audio_state.nb_hw_voices <= 0) { + dolog ("Bogus number of voices %d, resetting to 1\n", + audio_state.nb_hw_voices); + } + + drvname = audio_get_conf_str (QC_AUDIO_DRV, NULL); + if (drvname) { + int found = 0; + for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) { + if (!strcmp (drvname, drvtab[i]->name)) { + done = voice_init (drvtab[i]); + found = 1; + break; + } + } + if (!found) { + dolog ("Unknown audio driver `%s'\n", drvname); + } + } + + qemu_add_vm_stop_handler (audio_vm_stop_handler, NULL); + atexit (audio_atexit); + + if (!done) { + for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) { + if (drvtab[i]->can_be_default) + done = voice_init (drvtab[i]); + } + } + + audio_state.ticks_threshold = ticks_per_sec / 50; + audio_state.freq_threshold = 100; + + register_savevm ("audio", 0, 1, audio_save, audio_load, NULL); + if (!done) { + dolog ("Can not initialize audio subsystem\n"); + return; + } + + for (i = 0; hw_ctors[i]; i++) { + hw_ctors[i] (); + } +} diff --git a/audio/audio.h b/audio/audio.h new file mode 100644 index 0000000000..926a1bac93 --- /dev/null +++ b/audio/audio.h @@ -0,0 +1,188 @@ +/* + * QEMU Audio subsystem header + * + * Copyright (c) 2003-2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#ifndef QEMU_AUDIO_H +#define QEMU_AUDIO_H + +#include "mixeng.h" + +#define dolog(...) fprintf (stderr, AUDIO_CAP ": " __VA_ARGS__) +#ifdef DEBUG +#define ldebug(...) dolog (__VA_ARGS__) +#else +#define ldebug(...) +#endif + +typedef enum { + AUD_FMT_U8, + AUD_FMT_S8, + AUD_FMT_U16, + AUD_FMT_S16 +} audfmt_e; + +typedef struct HWVoice HWVoice; +struct audio_output_driver; + +typedef struct AudioState { + int fixed_format; + int fixed_freq; + int fixed_channels; + int fixed_fmt; + int nb_hw_voices; + int voice_size; + int64_t ticks_threshold; + int freq_threshold; + void *opaque; + struct audio_output_driver *drv; +} AudioState; + +extern AudioState audio_state; + +typedef struct SWVoice { + int freq; + audfmt_e fmt; + int nchannels; + + int shift; + int align; + + t_sample *conv; + + int left; + int pos; + int bytes_per_second; + int64_t ratio; + st_sample_t *buf; + void *rate; + + int wpos; + int live; + int active; + int64_t old_ticks; + HWVoice *hw; + char *name; +} SWVoice; + +#define VOICE_ENABLE 1 +#define VOICE_DISABLE 2 + +struct pcm_ops { + int (*init) (HWVoice *hw, int freq, int nchannels, audfmt_e fmt); + void (*fini) (HWVoice *hw); + void (*run) (HWVoice *hw); + int (*write) (SWVoice *sw, void *buf, int size); + int (*ctl) (HWVoice *hw, int cmd, ...); +}; + +struct audio_output_driver { + const char *name; + void *(*init) (void); + void (*fini) (void *); + struct pcm_ops *pcm_ops; + int can_be_default; + int max_voices; + int voice_size; +}; + +struct HWVoice { + int active; + int enabled; + int pending_disable; + int valid; + int freq; + + f_sample *clip; + audfmt_e fmt; + int nchannels; + + int align; + int shift; + + int rpos; + int bufsize; + + int bytes_per_second; + st_sample_t *mix_buf; + + int samples; + int64_t old_ticks; + int nb_voices; + struct SWVoice **pvoice; + struct pcm_ops *pcm_ops; +}; + +void audio_log (const char *fmt, ...); +void pcm_sw_free_resources (SWVoice *sw); +int pcm_sw_alloc_resources (SWVoice *sw); +void pcm_sw_fini (SWVoice *sw); +int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq, + int nchannels, audfmt_e fmt); + +void pcm_hw_clear (HWVoice *hw, void *buf, int len); +HWVoice * pcm_hw_find_any (HWVoice *hw); +HWVoice * pcm_hw_find_any_active (HWVoice *hw); +HWVoice * pcm_hw_find_any_passive (HWVoice *hw); +HWVoice * pcm_hw_find_specific (HWVoice *hw, int freq, + int nchannels, audfmt_e fmt); +HWVoice * pcm_hw_add (int freq, int nchannels, audfmt_e fmt); +int pcm_hw_add_sw (HWVoice *hw, SWVoice *sw); +int pcm_hw_del_sw (HWVoice *hw, SWVoice *sw); +SWVoice * pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt); + +void pcm_hw_free_resources (HWVoice *hw); +int pcm_hw_alloc_resources (HWVoice *hw); +void pcm_hw_fini (HWVoice *hw); +void pcm_hw_gc (HWVoice *hw); +int pcm_hw_get_live (HWVoice *hw); +int pcm_hw_get_live2 (HWVoice *hw, int *nb_active); +void pcm_hw_dec_live (HWVoice *hw, int decr); +int pcm_hw_write (SWVoice *sw, void *buf, int len); + +int audio_get_conf_int (const char *key, int defval); +const char *audio_get_conf_str (const char *key, const char *defval); + +/* Public API */ +SWVoice * AUD_open (SWVoice *sw, const char *name, int freq, + int nchannels, audfmt_e fmt); +int AUD_write (SWVoice *sw, void *pcm_buf, int size); +void AUD_adjust (SWVoice *sw, int leftover); +void AUD_reset (SWVoice *sw); +int AUD_get_free (SWVoice *sw); +int AUD_get_buffer_size (SWVoice *sw); +void AUD_run (void); +void AUD_enable (SWVoice *sw, int on); +int AUD_calc_elapsed (SWVoice *sw); + +static inline void *advance (void *p, int incr) +{ + uint8_t *d = p; + return (d + incr); +} + +uint32_t popcount (uint32_t u); +inline uint32_t lsbindex (uint32_t u); + +#define audio_MIN(a, b) ((a)>(b)?(b):(a)) +#define audio_MAX(a, b) ((a)<(b)?(b):(a)) + +#endif /* audio.h */ diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c new file mode 100644 index 0000000000..7457033f92 --- /dev/null +++ b/audio/fmodaudio.c @@ -0,0 +1,457 @@ +/* + * QEMU FMOD audio output driver + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include <fmod.h> +#include <fmod_errors.h> +#include "vl.h" + +#define AUDIO_CAP "fmod" +#include "audio/audio.h" +#include "audio/fmodaudio.h" + +#define QC_FMOD_DRV "QEMU_FMOD_DRV" +#define QC_FMOD_FREQ "QEMU_FMOD_FREQ" +#define QC_FMOD_SAMPLES "QEMU_FMOD_SAMPLES" +#define QC_FMOD_CHANNELS "QEMU_FMOD_CHANNELS" +#define QC_FMOD_BUFSIZE "QEMU_FMOD_BUFSIZE" +#define QC_FMOD_THRESHOLD "QEMU_FMOD_THRESHOLD" + +static struct { + int nb_samples; + int freq; + int nb_channels; + int bufsize; + int threshold; +} conf = { + 2048, + 44100, + 1, + 0, + 128 +}; + +#define errstr() FMOD_ErrorString (FSOUND_GetError ()) + +static int fmod_hw_write (SWVoice *sw, void *buf, int len) +{ + return pcm_hw_write (sw, buf, len); +} + +static void fmod_clear_sample (FMODVoice *fmd) +{ + HWVoice *hw = &fmd->hw; + int status; + void *p1 = 0, *p2 = 0; + unsigned int len1 = 0, len2 = 0; + + status = FSOUND_Sample_Lock ( + fmd->fmod_sample, + 0, + hw->samples << hw->shift, + &p1, + &p2, + &len1, + &len2 + ); + + if (!status) { + dolog ("Failed to lock sample\nReason: %s\n", errstr ()); + return; + } + + if ((len1 & hw->align) || (len2 & hw->align)) { + dolog ("Locking sample returned unaligned length %d, %d\n", + len1, len2); + goto fail; + } + + if (len1 + len2 != hw->samples << hw->shift) { + dolog ("Locking sample returned incomplete length %d, %d\n", + len1 + len2, hw->samples << hw->shift); + goto fail; + } + pcm_hw_clear (hw, p1, hw->samples); + + fail: + status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, len1, len2); + if (!status) { + dolog ("Failed to unlock sample\nReason: %s\n", errstr ()); + } +} + +static int fmod_write_sample (HWVoice *hw, uint8_t *dst, st_sample_t *src, + int src_size, int src_pos, int dst_len) +{ + int src_len1 = dst_len, src_len2 = 0, pos = src_pos + dst_len; + st_sample_t *src1 = src + src_pos, *src2 = 0; + + if (src_pos + dst_len > src_size) { + src_len1 = src_size - src_pos; + src2 = src; + src_len2 = dst_len - src_len1; + pos = src_len2; + } + + if (src_len1) { + hw->clip (dst, src1, src_len1); + memset (src1, 0, src_len1 * sizeof (st_sample_t)); + advance (dst, src_len1); + } + + if (src_len2) { + hw->clip (dst, src2, src_len2); + memset (src2, 0, src_len2 * sizeof (st_sample_t)); + } + return pos; +} + +static int fmod_unlock_sample (FMODVoice *fmd, void *p1, void *p2, + unsigned int blen1, unsigned int blen2) +{ + int status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, blen1, blen2); + if (!status) { + dolog ("Failed to unlock sample\nReason: %s\n", errstr ()); + return -1; + } + return 0; +} + +static int fmod_lock_sample (FMODVoice *fmd, int pos, int len, + void **p1, void **p2, + unsigned int *blen1, unsigned int *blen2) +{ + HWVoice *hw = &fmd->hw; + int status; + + status = FSOUND_Sample_Lock ( + fmd->fmod_sample, + pos << hw->shift, + len << hw->shift, + p1, + p2, + blen1, + blen2 + ); + + if (!status) { + dolog ("Failed to lock sample\nReason: %s\n", errstr ()); + return -1; + } + + if ((*blen1 & hw->align) || (*blen2 & hw->align)) { + dolog ("Locking sample returned unaligned length %d, %d\n", + *blen1, *blen2); + fmod_unlock_sample (fmd, *p1, *p2, *blen1, *blen2); + return -1; + } + return 0; +} + +static void fmod_hw_run (HWVoice *hw) +{ + FMODVoice *fmd = (FMODVoice *) hw; + int rpos, live, decr; + void *p1 = 0, *p2 = 0; + unsigned int blen1 = 0, blen2 = 0; + unsigned int len1 = 0, len2 = 0; + int nb_active; + + live = pcm_hw_get_live2 (hw, &nb_active); + if (live <= 0) { + return; + } + + if (!hw->pending_disable + && nb_active + && conf.threshold + && live <= conf.threshold) { + ldebug ("live=%d nb_active=%d\n", live, nb_active); + return; + } + + decr = live; + +#if 1 + if (fmd->channel >= 0) { + int pos2 = (fmd->old_pos + decr) % hw->samples; + int pos = FSOUND_GetCurrentPosition (fmd->channel); + + if (fmd->old_pos < pos && pos2 >= pos) { + decr = pos - fmd->old_pos - (pos2 == pos) - 1; + } + else if (fmd->old_pos > pos && pos2 >= pos && pos2 < fmd->old_pos) { + decr = (hw->samples - fmd->old_pos) + pos - (pos2 == pos) - 1; + } +/* ldebug ("pos=%d pos2=%d old=%d live=%d decr=%d\n", */ +/* pos, pos2, fmd->old_pos, live, decr); */ + } +#endif + + if (decr <= 0) { + return; + } + + if (fmod_lock_sample (fmd, fmd->old_pos, decr, &p1, &p2, &blen1, &blen2)) { + return; + } + + len1 = blen1 >> hw->shift; + len2 = blen2 >> hw->shift; + ldebug ("%p %p %d %d %d %d\n", p1, p2, len1, len2, blen1, blen2); + decr = len1 + len2; + rpos = hw->rpos; + + if (len1) { + rpos = fmod_write_sample (hw, p1, hw->mix_buf, hw->samples, rpos, len1); + } + + if (len2) { + rpos = fmod_write_sample (hw, p2, hw->mix_buf, hw->samples, rpos, len2); + } + + fmod_unlock_sample (fmd, p1, p2, blen1, blen2); + + pcm_hw_dec_live (hw, decr); + hw->rpos = rpos % hw->samples; + fmd->old_pos = (fmd->old_pos + decr) % hw->samples; +} + +static int AUD_to_fmodfmt (audfmt_e fmt, int stereo) +{ + int mode = FSOUND_LOOP_NORMAL; + + switch (fmt) { + case AUD_FMT_S8: + mode |= FSOUND_SIGNED | FSOUND_8BITS; + break; + + case AUD_FMT_U8: + mode |= FSOUND_UNSIGNED | FSOUND_8BITS; + break; + + case AUD_FMT_S16: + mode |= FSOUND_SIGNED | FSOUND_16BITS; + break; + + case AUD_FMT_U16: + mode |= FSOUND_UNSIGNED | FSOUND_16BITS; + break; + + default: + dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt); + exit (EXIT_FAILURE); + } + mode |= stereo ? FSOUND_STEREO : FSOUND_MONO; + return mode; +} + +static void fmod_hw_fini (HWVoice *hw) +{ + FMODVoice *fmd = (FMODVoice *) hw; + + if (fmd->fmod_sample) { + FSOUND_Sample_Free (fmd->fmod_sample); + fmd->fmod_sample = 0; + + if (fmd->channel >= 0) { + FSOUND_StopSound (fmd->channel); + } + } +} + +static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt) +{ + int bits16, mode, channel; + FMODVoice *fmd = (FMODVoice *) hw; + + mode = AUD_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0); + fmd->fmod_sample = FSOUND_Sample_Alloc ( + FSOUND_FREE, /* index */ + conf.nb_samples, /* length */ + mode, /* mode */ + freq, /* freq */ + 255, /* volume */ + 128, /* pan */ + 255 /* priority */ + ); + + if (!fmd->fmod_sample) { + dolog ("Failed to allocate FMOD sample\nReason: %s\n", errstr ()); + return -1; + } + + channel = FSOUND_PlaySoundEx (FSOUND_FREE, fmd->fmod_sample, 0, 1); + if (channel < 0) { + dolog ("Failed to start playing sound\nReason: %s\n", errstr ()); + FSOUND_Sample_Free (fmd->fmod_sample); + return -1; + } + fmd->channel = channel; + + hw->freq = freq; + hw->fmt = fmt; + hw->nchannels = nchannels; + bits16 = fmt == AUD_FMT_U16 || fmt == AUD_FMT_S16; + hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16; + return 0; +} + +static int fmod_hw_ctl (HWVoice *hw, int cmd, ...) +{ + int status; + FMODVoice *fmd = (FMODVoice *) hw; + + switch (cmd) { + case VOICE_ENABLE: + fmod_clear_sample (fmd); + status = FSOUND_SetPaused (fmd->channel, 0); + if (!status) { + dolog ("Failed to resume channel %d\nReason: %s\n", + fmd->channel, errstr ()); + } + break; + + case VOICE_DISABLE: + status = FSOUND_SetPaused (fmd->channel, 1); + if (!status) { + dolog ("Failed to pause channel %d\nReason: %s\n", + fmd->channel, errstr ()); + } + break; + } + return 0; +} + +static struct { + const char *name; + int type; +} drvtab[] = { + {"none", FSOUND_OUTPUT_NOSOUND}, +#ifdef _WIN32 + {"winmm", FSOUND_OUTPUT_WINMM}, + {"dsound", FSOUND_OUTPUT_DSOUND}, + {"a3d", FSOUND_OUTPUT_A3D}, + {"asio", FSOUND_OUTPUT_ASIO}, +#endif +#ifdef __linux__ + {"oss", FSOUND_OUTPUT_OSS}, + {"alsa", FSOUND_OUTPUT_ALSA}, + {"esd", FSOUND_OUTPUT_ESD}, +#endif +#ifdef __APPLE__ + {"mac", FSOUND_OUTPUT_MAC}, +#endif +#if 0 + {"xbox", FSOUND_OUTPUT_XBOX}, + {"ps2", FSOUND_OUTPUT_PS2}, + {"gcube", FSOUND_OUTPUT_GC}, +#endif + {"nort", FSOUND_OUTPUT_NOSOUND_NONREALTIME} +}; + +static void *fmod_audio_init (void) +{ + int i; + double ver; + int status; + int output_type = -1; + const char *drv = audio_get_conf_str (QC_FMOD_DRV, NULL); + + ver = FSOUND_GetVersion (); + if (ver < FMOD_VERSION) { + dolog ("Wrong FMOD version %f, need at least %f\n", ver, FMOD_VERSION); + return NULL; + } + + if (drv) { + int found = 0; + for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) { + if (!strcmp (drv, drvtab[i].name)) { + output_type = drvtab[i].type; + found = 1; + break; + } + } + if (!found) { + dolog ("Unknown FMOD output driver `%s'\n", drv); + } + } + + if (output_type != -1) { + status = FSOUND_SetOutput (output_type); + if (!status) { + dolog ("FSOUND_SetOutput(%d) failed\nReason: %s\n", + output_type, errstr ()); + return NULL; + } + } + + conf.freq = audio_get_conf_int (QC_FMOD_FREQ, conf.freq); + conf.nb_samples = audio_get_conf_int (QC_FMOD_SAMPLES, conf.nb_samples); + conf.nb_channels = + audio_get_conf_int (QC_FMOD_CHANNELS, + (audio_state.nb_hw_voices > 1 + ? audio_state.nb_hw_voices + : conf.nb_channels)); + conf.bufsize = audio_get_conf_int (QC_FMOD_BUFSIZE, conf.bufsize); + conf.threshold = audio_get_conf_int (QC_FMOD_THRESHOLD, conf.threshold); + + if (conf.bufsize) { + status = FSOUND_SetBufferSize (conf.bufsize); + if (!status) { + dolog ("FSOUND_SetBufferSize (%d) failed\nReason: %s\n", + conf.bufsize, errstr ()); + } + } + + status = FSOUND_Init (conf.freq, conf.nb_channels, 0); + if (!status) { + dolog ("FSOUND_Init failed\nReason: %s\n", errstr ()); + return NULL; + } + + return &conf; +} + +static void fmod_audio_fini (void *opaque) +{ + FSOUND_Close (); +} + +struct pcm_ops fmod_pcm_ops = { + fmod_hw_init, + fmod_hw_fini, + fmod_hw_run, + fmod_hw_write, + fmod_hw_ctl +}; + +struct audio_output_driver fmod_output_driver = { + "fmod", + fmod_audio_init, + fmod_audio_fini, + &fmod_pcm_ops, + 1, + INT_MAX, + sizeof (FMODVoice) +}; diff --git a/audio/fmodaudio.h b/audio/fmodaudio.h new file mode 100644 index 0000000000..9f85c30804 --- /dev/null +++ b/audio/fmodaudio.h @@ -0,0 +1,39 @@ +/* + * QEMU FMOD audio output driver header + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#ifndef QEMU_FMODAUDIO_H +#define QEMU_FMODAUDIO_H + +#include <fmod.h> + +typedef struct FMODVoice { + struct HWVoice hw; + unsigned int old_pos; + FSOUND_SAMPLE *fmod_sample; + int channel; +} FMODVoice; + +extern struct pcm_ops fmod_pcm_ops; +extern struct audio_output_driver fmod_output_driver; + +#endif /* fmodaudio.h */ diff --git a/audio/mixeng.c b/audio/mixeng.c new file mode 100644 index 0000000000..b0bb412c63 --- /dev/null +++ b/audio/mixeng.c @@ -0,0 +1,255 @@ +/* + * QEMU Mixing engine + * + * Copyright (c) 2004 Vassili Karpov (malc) + * Copyright (c) 1998 Fabrice Bellard + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include "vl.h" +//#define DEBUG_FP +#include "audio/mixeng.h" + +#define IN_T int8_t +#define IN_MIN CHAR_MIN +#define IN_MAX CHAR_MAX +#define SIGNED +#include "mixeng_template.h" +#undef SIGNED +#undef IN_MAX +#undef IN_MIN +#undef IN_T + +#define IN_T uint8_t +#define IN_MIN 0 +#define IN_MAX UCHAR_MAX +#include "mixeng_template.h" +#undef IN_MAX +#undef IN_MIN +#undef IN_T + +#define IN_T int16_t +#define IN_MIN SHRT_MIN +#define IN_MAX SHRT_MAX +#define SIGNED +#include "mixeng_template.h" +#undef SIGNED +#undef IN_MAX +#undef IN_MIN +#undef IN_T + +#define IN_T uint16_t +#define IN_MIN 0 +#define IN_MAX USHRT_MAX +#include "mixeng_template.h" +#undef IN_MAX +#undef IN_MIN +#undef IN_T + +t_sample *mixeng_conv[2][2][2] = { + { + { + conv_uint8_t_to_mono, + conv_uint16_t_to_mono + }, + { + conv_int8_t_to_mono, + conv_int16_t_to_mono + } + }, + { + { + conv_uint8_t_to_stereo, + conv_uint16_t_to_stereo + }, + { + conv_int8_t_to_stereo, + conv_int16_t_to_stereo + } + } +}; + +f_sample *mixeng_clip[2][2][2] = { + { + { + clip_uint8_t_from_mono, + clip_uint16_t_from_mono + }, + { + clip_int8_t_from_mono, + clip_int16_t_from_mono + } + }, + { + { + clip_uint8_t_from_stereo, + clip_uint16_t_from_stereo + }, + { + clip_int8_t_from_stereo, + clip_int16_t_from_stereo + } + } +}; + +/* + * August 21, 1998 + * Copyright 1998 Fabrice Bellard. + * + * [Rewrote completly the code of Lance Norskog And Sundry + * Contributors with a more efficient algorithm.] + * + * This source code is freely redistributable and may be used for + * any purpose. This copyright notice must be maintained. + * Lance Norskog And Sundry Contributors are not responsible for + * the consequences of using this software. + */ + +/* + * Sound Tools rate change effect file. + */ +/* + * Linear Interpolation. + * + * The use of fractional increment allows us to use no buffer. It + * avoid the problems at the end of the buffer we had with the old + * method which stored a possibly big buffer of size + * lcm(in_rate,out_rate). + * + * Limited to 16 bit samples and sampling frequency <= 65535 Hz. If + * the input & output frequencies are equal, a delay of one sample is + * introduced. Limited to processing 32-bit count worth of samples. + * + * 1 << FRAC_BITS evaluating to zero in several places. Changed with + * an (unsigned long) cast to make it safe. MarkMLl 2/1/99 + */ + +/* Private data */ +typedef struct ratestuff { + uint64_t opos; + uint64_t opos_inc; + uint32_t ipos; /* position in the input stream (integer) */ + st_sample_t ilast; /* last sample in the input stream */ +} *rate_t; + +/* + * Prepare processing. + */ +void *st_rate_start (int inrate, int outrate) +{ + rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff)); + + if (!rate) { + exit (EXIT_FAILURE); + } + + if (inrate == outrate) { + // exit (EXIT_FAILURE); + } + + if (inrate >= 65535 || outrate >= 65535) { + // exit (EXIT_FAILURE); + } + + rate->opos = 0; + + /* increment */ + rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate; + + rate->ipos = 0; + rate->ilast.l = 0; + rate->ilast.r = 0; + return rate; +} + +/* + * Processed signed long samples from ibuf to obuf. + * Return number of samples processed. + */ +void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf, + int *isamp, int *osamp) +{ + rate_t rate = (rate_t) opaque; + st_sample_t *istart, *iend; + st_sample_t *ostart, *oend; + st_sample_t ilast, icur, out; + int64_t t; + + ilast = rate->ilast; + + istart = ibuf; + iend = ibuf + *isamp; + + ostart = obuf; + oend = obuf + *osamp; + + if (rate->opos_inc == 1ULL << 32) { + int i, n = *isamp > *osamp ? *osamp : *isamp; + for (i = 0; i < n; i++) { + obuf[i].l += ibuf[i].r; + obuf[i].r += ibuf[i].r; + } + *isamp = n; + *osamp = n; + return; + } + + while (obuf < oend) { + + /* Safety catch to make sure we have input samples. */ + if (ibuf >= iend) + break; + + /* read as many input samples so that ipos > opos */ + + while (rate->ipos <= (rate->opos >> 32)) { + ilast = *ibuf++; + rate->ipos++; + /* See if we finished the input buffer yet */ + if (ibuf >= iend) goto the_end; + } + + icur = *ibuf; + + /* interpolate */ + t = rate->opos & 0xffffffff; + out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX; + out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX; + + /* output sample & increment position */ +#if 0 + *obuf++ = out; +#else + obuf->l += out.l; + obuf->r += out.r; + obuf += 1; +#endif + rate->opos += rate->opos_inc; + } + +the_end: + *isamp = ibuf - istart; + *osamp = obuf - ostart; + rate->ilast = ilast; +} + +void st_rate_stop (void *opaque) +{ + qemu_free (opaque); +} diff --git a/audio/mixeng.h b/audio/mixeng.h new file mode 100644 index 0000000000..699435ea25 --- /dev/null +++ b/audio/mixeng.h @@ -0,0 +1,39 @@ +/* + * QEMU Mixing engine header + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#ifndef QEMU_MIXENG_H +#define QEMU_MIXENG_H + +typedef void (t_sample) (void *dst, const void *src, int samples); +typedef void (f_sample) (void *dst, const void *src, int samples); +typedef struct { int64_t l; int64_t r; } st_sample_t; + +extern t_sample *mixeng_conv[2][2][2]; +extern f_sample *mixeng_clip[2][2][2]; + +void *st_rate_start (int inrate, int outrate); +void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf, + int *isamp, int *osamp); +void st_rate_stop (void *opaque); + +#endif /* mixeng.h */ diff --git a/audio/mixeng_template.h b/audio/mixeng_template.h new file mode 100644 index 0000000000..f3b3f654fd --- /dev/null +++ b/audio/mixeng_template.h @@ -0,0 +1,111 @@ +/* + * QEMU Mixing engine + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/* + * Tusen tack till Mike Nordell + * dec++'ified by Dscho + */ + +#ifdef SIGNED +#define HALFT IN_MAX +#define HALF IN_MAX +#else +#define HALFT ((IN_MAX)>>1) +#define HALF HALFT +#endif + +static int64_t inline glue(conv_,IN_T) (IN_T v) +{ +#ifdef SIGNED + return (INT_MAX*(int64_t)v)/HALF; +#else + return (INT_MAX*((int64_t)v-HALFT))/HALF; +#endif +} + +static IN_T inline glue(clip_,IN_T) (int64_t v) +{ + if (v >= INT_MAX) + return IN_MAX; + else if (v < -INT_MAX) + return IN_MIN; + +#ifdef SIGNED + return (IN_T) (v*HALF/INT_MAX); +#else + return (IN_T) (v+INT_MAX/2)*HALF/INT_MAX; +#endif +} + +static void glue(glue(conv_,IN_T),_to_stereo) (void *dst, const void *src, + int samples) +{ + st_sample_t *out = (st_sample_t *) dst; + IN_T *in = (IN_T *) src; + while (samples--) { + out->l = glue(conv_,IN_T) (*in++); + out->r = glue(conv_,IN_T) (*in++); + out += 1; + } +} + +static void glue(glue(conv_,IN_T),_to_mono) (void *dst, const void *src, + int samples) +{ + st_sample_t *out = (st_sample_t *) dst; + IN_T *in = (IN_T *) src; + while (samples--) { + out->l = glue(conv_,IN_T) (in[0]); + out->r = out->l; + out += 1; + in += 1; + } +} + +static void glue(glue(clip_,IN_T),_from_stereo) (void *dst, const void *src, + int samples) +{ + st_sample_t *in = (st_sample_t *) src; + IN_T *out = (IN_T *) dst; + while (samples--) { + *out++ = glue(clip_,IN_T) (in->l); + *out++ = glue(clip_,IN_T) (in->r); + in += 1; + } +} + +static void glue(glue(clip_,IN_T),_from_mono) (void *dst, const void *src, + int samples) +{ + st_sample_t *in = (st_sample_t *) src; + IN_T *out = (IN_T *) dst; + while (samples--) { + *out++ = glue(clip_,IN_T) (in->l + in->r); + in += 1; + } +} + +#undef HALF +#undef HALFT + diff --git a/audio/ossaudio.c b/audio/ossaudio.c new file mode 100644 index 0000000000..9fefaa3a27 --- /dev/null +++ b/audio/ossaudio.c @@ -0,0 +1,466 @@ +/* + * QEMU OSS audio output driver + * + * Copyright (c) 2003-2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/* Temporary kludge */ +#if defined __linux__ || (defined _BSD && !defined __APPLE__) +#include <assert.h> +#include "vl.h" + +#include <sys/mman.h> +#include <sys/types.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> + +#define AUDIO_CAP "oss" +#include "audio/audio.h" +#include "audio/ossaudio.h" + +#define QC_OSS_FRAGSIZE "QEMU_OSS_FRAGSIZE" +#define QC_OSS_NFRAGS "QEMU_OSS_NFRAGS" +#define QC_OSS_MMAP "QEMU_OSS_MMAP" +#define QC_OSS_DEV "QEMU_OSS_DEV" + +#define errstr() strerror (errno) + +static struct { + int try_mmap; + int nfrags; + int fragsize; + const char *dspname; +} conf = { + .try_mmap = 0, + .nfrags = 4, + .fragsize = 4096, + .dspname = "/dev/dsp" +}; + +struct oss_params { + int freq; + audfmt_e fmt; + int nchannels; + int nfrags; + int fragsize; +}; + +static int oss_hw_write (SWVoice *sw, void *buf, int len) +{ + return pcm_hw_write (sw, buf, len); +} + +static int AUD_to_ossfmt (audfmt_e fmt) +{ + switch (fmt) { + case AUD_FMT_S8: return AFMT_S8; + case AUD_FMT_U8: return AFMT_U8; + case AUD_FMT_S16: return AFMT_S16_LE; + case AUD_FMT_U16: return AFMT_U16_LE; + default: + dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt); + exit (EXIT_FAILURE); + } +} + +static int oss_to_audfmt (int fmt) +{ + switch (fmt) { + case AFMT_S8: return AUD_FMT_S8; + case AFMT_U8: return AUD_FMT_U8; + case AFMT_S16_LE: return AUD_FMT_S16; + case AFMT_U16_LE: return AUD_FMT_U16; + default: + dolog ("Internal logic error: Unrecognized OSS audio format %d\n" + "Aborting\n", + fmt); + exit (EXIT_FAILURE); + } +} + +#ifdef DEBUG_PCM +static void oss_dump_pcm_info (struct oss_params *req, struct oss_params *obt) +{ + dolog ("parameter | requested value | obtained value\n"); + dolog ("format | %10d | %10d\n", req->fmt, obt->fmt); + dolog ("channels | %10d | %10d\n", req->nchannels, obt->nchannels); + dolog ("frequency | %10d | %10d\n", req->freq, obt->freq); + dolog ("nfrags | %10d | %10d\n", req->nfrags, obt->nfrags); + dolog ("fragsize | %10d | %10d\n", req->fragsize, obt->fragsize); +} +#endif + +static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd) +{ + int fd; + int mmmmssss; + audio_buf_info abinfo; + int fmt, freq, nchannels; + const char *dspname = conf.dspname; + + fd = open (dspname, O_RDWR | O_NONBLOCK); + if (-1 == fd) { + dolog ("Could not initialize audio hardware. Failed to open `%s':\n" + "Reason:%s\n", + dspname, + errstr ()); + return -1; + } + + freq = req->freq; + nchannels = req->nchannels; + fmt = req->fmt; + + if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) { + dolog ("Could not initialize audio hardware\n" + "Failed to set sample size\n" + "Reason: %s\n", + errstr ()); + goto err; + } + + if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) { + dolog ("Could not initialize audio hardware\n" + "Failed to set number of channels\n" + "Reason: %s\n", + errstr ()); + goto err; + } + + if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) { + dolog ("Could not initialize audio hardware\n" + "Failed to set frequency\n" + "Reason: %s\n", + errstr ()); + goto err; + } + + if (ioctl (fd, SNDCTL_DSP_NONBLOCK)) { + dolog ("Could not initialize audio hardware\n" + "Failed to set non-blocking mode\n" + "Reason: %s\n", + errstr ()); + goto err; + } + + mmmmssss = (req->nfrags << 16) | lsbindex (req->fragsize); + if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) { + dolog ("Could not initialize audio hardware\n" + "Failed to set buffer length (%d, %d)\n" + "Reason:%s\n", + conf.nfrags, conf.fragsize, + errstr ()); + goto err; + } + + if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &abinfo)) { + dolog ("Could not initialize audio hardware\n" + "Failed to get buffer length\n" + "Reason:%s\n", + errstr ()); + goto err; + } + + obt->fmt = fmt; + obt->nchannels = nchannels; + obt->freq = freq; + obt->nfrags = abinfo.fragstotal; + obt->fragsize = abinfo.fragsize; + *pfd = fd; + + if ((req->fmt != obt->fmt) || + (req->nchannels != obt->nchannels) || + (req->freq != obt->freq) || + (req->fragsize != obt->fragsize) || + (req->nfrags != obt->nfrags)) { +#ifdef DEBUG_PCM + dolog ("Audio parameters mismatch\n"); + oss_dump_pcm_info (req, obt); +#endif + } + +#ifdef DEBUG_PCM + oss_dump_pcm_info (req, obt); +#endif + return 0; + +err: + close (fd); + return -1; +} + +static void oss_hw_run (HWVoice *hw) +{ + OSSVoice *oss = (OSSVoice *) hw; + int err, rpos, live, decr; + int samples; + uint8_t *dst; + st_sample_t *src; + struct audio_buf_info abinfo; + struct count_info cntinfo; + + live = pcm_hw_get_live (hw); + if (live <= 0) + return; + + if (oss->mmapped) { + int bytes; + + err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo); + if (err < 0) { + dolog ("SNDCTL_DSP_GETOPTR failed\nReason: %s\n", errstr ()); + return; + } + + if (cntinfo.ptr == oss->old_optr) { + if (abs (hw->samples - live) < 64) + dolog ("overrun\n"); + return; + } + + if (cntinfo.ptr > oss->old_optr) { + bytes = cntinfo.ptr - oss->old_optr; + } + else { + bytes = hw->bufsize + cntinfo.ptr - oss->old_optr; + } + + decr = audio_MIN (bytes >> hw->shift, live); + } + else { + err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo); + if (err < 0) { + dolog ("SNDCTL_DSP_GETOSPACE failed\nReason: %s\n", errstr ()); + return; + } + + decr = audio_MIN (abinfo.bytes >> hw->shift, live); + if (decr <= 0) + return; + } + + samples = decr; + rpos = hw->rpos; + while (samples) { + int left_till_end_samples = hw->samples - rpos; + int convert_samples = audio_MIN (samples, left_till_end_samples); + + src = advance (hw->mix_buf, rpos * sizeof (st_sample_t)); + dst = advance (oss->pcm_buf, rpos << hw->shift); + + hw->clip (dst, src, convert_samples); + if (!oss->mmapped) { + int written; + + written = write (oss->fd, dst, convert_samples << hw->shift); + /* XXX: follow errno recommendations ? */ + if (written == -1) { + dolog ("Failed to write audio\nReason: %s\n", errstr ()); + continue; + } + + if (written != convert_samples << hw->shift) { + int wsamples = written >> hw->shift; + int wbytes = wsamples << hw->shift; + if (wbytes != written) { + dolog ("Unaligned write %d, %d\n", wbytes, written); + } + memset (src, 0, wbytes); + decr -= samples; + rpos = (rpos + wsamples) % hw->samples; + break; + } + } + memset (src, 0, convert_samples * sizeof (st_sample_t)); + + rpos = (rpos + convert_samples) % hw->samples; + samples -= convert_samples; + } + if (oss->mmapped) { + oss->old_optr = cntinfo.ptr; + } + + pcm_hw_dec_live (hw, decr); + hw->rpos = rpos; +} + +static void oss_hw_fini (HWVoice *hw) +{ + int err; + OSSVoice *oss = (OSSVoice *) hw; + + ldebug ("oss_hw_fini\n"); + err = close (oss->fd); + if (err) { + dolog ("Failed to close OSS descriptor\nReason: %s\n", errstr ()); + } + oss->fd = -1; + + if (oss->pcm_buf) { + if (oss->mmapped) { + err = munmap (oss->pcm_buf, hw->bufsize); + if (err) { + dolog ("Failed to unmap OSS buffer\nReason: %s\n", + errstr ()); + } + } + else { + qemu_free (oss->pcm_buf); + } + oss->pcm_buf = NULL; + } +} + +static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt) +{ + OSSVoice *oss = (OSSVoice *) hw; + struct oss_params req, obt; + + assert (!oss->fd); + req.fmt = AUD_to_ossfmt (fmt); + req.freq = freq; + req.nchannels = nchannels; + req.fragsize = conf.fragsize; + req.nfrags = conf.nfrags; + + if (oss_open (&req, &obt, &oss->fd)) + return -1; + + hw->freq = obt.freq; + hw->fmt = oss_to_audfmt (obt.fmt); + hw->nchannels = obt.nchannels; + + oss->nfrags = obt.nfrags; + oss->fragsize = obt.fragsize; + hw->bufsize = obt.nfrags * obt.fragsize; + + oss->mmapped = 0; + if (conf.try_mmap) { + oss->pcm_buf = mmap (0, hw->bufsize, PROT_READ | PROT_WRITE, + MAP_SHARED, oss->fd, 0); + if (oss->pcm_buf == MAP_FAILED) { + dolog ("Failed to mmap OSS device\nReason: %s\n", + errstr ()); + } + + for (;;) { + int err; + int trig = 0; + if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) { + dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n", + errstr ()); + goto fail; + } + + trig = PCM_ENABLE_OUTPUT; + if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) { + dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n" + "Reason: %s\n", errstr ()); + goto fail; + } + oss->mmapped = 1; + break; + + fail: + err = munmap (oss->pcm_buf, hw->bufsize); + if (err) { + dolog ("Failed to unmap OSS device\nReason: %s\n", + errstr ()); + } + } + } + + if (!oss->mmapped) { + oss->pcm_buf = qemu_mallocz (hw->bufsize); + if (!oss->pcm_buf) { + close (oss->fd); + oss->fd = -1; + return -1; + } + } + + return 0; +} + +static int oss_hw_ctl (HWVoice *hw, int cmd, ...) +{ + int trig; + OSSVoice *oss = (OSSVoice *) hw; + + if (!oss->mmapped) + return 0; + + switch (cmd) { + case VOICE_ENABLE: + ldebug ("enabling voice\n"); + pcm_hw_clear (hw, oss->pcm_buf, hw->samples); + trig = PCM_ENABLE_OUTPUT; + if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) { + dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n" + "Reason: %s\n", errstr ()); + return -1; + } + break; + + case VOICE_DISABLE: + ldebug ("disabling voice\n"); + trig = 0; + if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) { + dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n", + errstr ()); + return -1; + } + break; + } + return 0; +} + +static void *oss_audio_init (void) +{ + conf.fragsize = audio_get_conf_int (QC_OSS_FRAGSIZE, conf.fragsize); + conf.nfrags = audio_get_conf_int (QC_OSS_NFRAGS, conf.nfrags); + conf.try_mmap = audio_get_conf_int (QC_OSS_MMAP, conf.try_mmap); + conf.dspname = audio_get_conf_str (QC_OSS_DEV, conf.dspname); + return &conf; +} + +static void oss_audio_fini (void *opaque) +{ +} + +struct pcm_ops oss_pcm_ops = { + oss_hw_init, + oss_hw_fini, + oss_hw_run, + oss_hw_write, + oss_hw_ctl +}; + +struct audio_output_driver oss_output_driver = { + "oss", + oss_audio_init, + oss_audio_fini, + &oss_pcm_ops, + 1, + INT_MAX, + sizeof (OSSVoice) +}; +#endif diff --git a/audio/ossaudio.h b/audio/ossaudio.h new file mode 100644 index 0000000000..f7d3ebd522 --- /dev/null +++ b/audio/ossaudio.h @@ -0,0 +1,40 @@ +/* + * QEMU OSS audio output driver header + * + * Copyright (c) 2003-2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#ifndef QEMU_OSSAUDIO_H +#define QEMU_OSSAUDIO_H + +typedef struct OSSVoice { + struct HWVoice hw; + void *pcm_buf; + int fd; + int nfrags; + int fragsize; + int mmapped; + int old_optr; +} OSSVoice; + +extern struct pcm_ops oss_pcm_ops; +extern struct audio_output_driver oss_output_driver; + +#endif /* ossaudio.h */ diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c new file mode 100644 index 0000000000..4d75853422 --- /dev/null +++ b/audio/sdlaudio.c @@ -0,0 +1,323 @@ +/* + * QEMU SDL audio output driver + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include <SDL/SDL.h> +#include <SDL/SDL_thread.h> +#include "vl.h" + +#define AUDIO_CAP "sdl" +#include "audio/audio.h" +#include "audio/sdlaudio.h" + +#define QC_SDL_SAMPLES "QEMU_SDL_SAMPLES" + +#define errstr() SDL_GetError () + +static struct { + int nb_samples; +} conf = { + 1024 +}; + +struct SDLAudioState { + int exit; + SDL_mutex *mutex; + SDL_sem *sem; + int initialized; +} glob_sdl; +typedef struct SDLAudioState SDLAudioState; + +static void sdl_hw_run (HWVoice *hw) +{ + (void) hw; +} + +static int sdl_lock (SDLAudioState *s) +{ + if (SDL_LockMutex (s->mutex)) { + dolog ("SDL_LockMutex failed\nReason: %s\n", errstr ()); + return -1; + } + return 0; +} + +static int sdl_unlock (SDLAudioState *s) +{ + if (SDL_UnlockMutex (s->mutex)) { + dolog ("SDL_UnlockMutex failed\nReason: %s\n", errstr ()); + return -1; + } + return 0; +} + +static int sdl_post (SDLAudioState *s) +{ + if (SDL_SemPost (s->sem)) { + dolog ("SDL_SemPost failed\nReason: %s\n", errstr ()); + return -1; + } + return 0; +} + +static int sdl_wait (SDLAudioState *s) +{ + if (SDL_SemWait (s->sem)) { + dolog ("SDL_SemWait failed\nReason: %s\n", errstr ()); + return -1; + } + return 0; +} + +static int sdl_unlock_and_post (SDLAudioState *s) +{ + if (sdl_unlock (s)) + return -1; + + return sdl_post (s); +} + +static int sdl_hw_write (SWVoice *sw, void *buf, int len) +{ + int ret; + SDLAudioState *s = &glob_sdl; + sdl_lock (s); + ret = pcm_hw_write (sw, buf, len); + sdl_unlock_and_post (s); + return ret; +} + +static int AUD_to_sdlfmt (audfmt_e fmt, int *shift) +{ + *shift = 0; + switch (fmt) { + case AUD_FMT_S8: return AUDIO_S8; + case AUD_FMT_U8: return AUDIO_U8; + case AUD_FMT_S16: *shift = 1; return AUDIO_S16LSB; + case AUD_FMT_U16: *shift = 1; return AUDIO_U16LSB; + default: + dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt); + exit (EXIT_FAILURE); + } +} + +static int sdl_to_audfmt (int fmt) +{ + switch (fmt) { + case AUDIO_S8: return AUD_FMT_S8; + case AUDIO_U8: return AUD_FMT_U8; + case AUDIO_S16LSB: return AUD_FMT_S16; + case AUDIO_U16LSB: return AUD_FMT_U16; + default: + dolog ("Internal logic error: Unrecognized SDL audio format %d\n" + "Aborting\n", fmt); + exit (EXIT_FAILURE); + } +} + +static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt) +{ + int status; + + status = SDL_OpenAudio (req, obt); + if (status) { + dolog ("SDL_OpenAudio failed\nReason: %s\n", errstr ()); + } + return status; +} + +static void sdl_close (SDLAudioState *s) +{ + if (s->initialized) { + sdl_lock (s); + s->exit = 1; + sdl_unlock_and_post (s); + SDL_PauseAudio (1); + SDL_CloseAudio (); + s->initialized = 0; + } +} + +static void sdl_callback (void *opaque, Uint8 *buf, int len) +{ + SDLVoice *sdl = opaque; + SDLAudioState *s = &glob_sdl; + HWVoice *hw = &sdl->hw; + int samples = len >> hw->shift; + + if (s->exit) { + return; + } + + while (samples) { + int to_mix, live, decr; + + /* dolog ("in callback samples=%d\n", samples); */ + sdl_wait (s); + if (s->exit) { + return; + } + + sdl_lock (s); + live = pcm_hw_get_live (hw); + if (live <= 0) + goto again; + + /* dolog ("in callback live=%d\n", live); */ + to_mix = audio_MIN (samples, live); + decr = to_mix; + while (to_mix) { + int chunk = audio_MIN (to_mix, hw->samples - hw->rpos); + st_sample_t *src = hw->mix_buf + hw->rpos; + + /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */ + hw->clip (buf, src, chunk); + memset (src, 0, chunk * sizeof (st_sample_t)); + hw->rpos = (hw->rpos + chunk) % hw->samples; + to_mix -= chunk; + buf += chunk << hw->shift; + } + samples -= decr; + pcm_hw_dec_live (hw, decr); + + again: + sdl_unlock (s); + } + /* dolog ("done len=%d\n", len); */ +} + +static void sdl_hw_fini (HWVoice *hw) +{ + ldebug ("sdl_hw_fini %d fixed=%d\n", + glob_sdl.initialized, audio_conf.fixed_format); + sdl_close (&glob_sdl); +} + +static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt) +{ + SDLVoice *sdl = (SDLVoice *) hw; + SDLAudioState *s = &glob_sdl; + SDL_AudioSpec req, obt; + int shift; + + ldebug ("sdl_hw_init %d freq=%d fixed=%d\n", + s->initialized, freq, audio_conf.fixed_format); + + if (nchannels != 2) { + dolog ("Bogus channel count %d\n", nchannels); + return -1; + } + + req.freq = freq; + req.format = AUD_to_sdlfmt (fmt, &shift); + req.channels = nchannels; + req.samples = conf.nb_samples; + shift <<= nchannels == 2; + + req.callback = sdl_callback; + req.userdata = sdl; + + if (sdl_open (&req, &obt)) + return -1; + + hw->freq = obt.freq; + hw->fmt = sdl_to_audfmt (obt.format); + hw->nchannels = obt.channels; + hw->bufsize = obt.samples << shift; + + s->initialized = 1; + s->exit = 0; + SDL_PauseAudio (0); + return 0; +} + +static int sdl_hw_ctl (HWVoice *hw, int cmd, ...) +{ + (void) hw; + + switch (cmd) { + case VOICE_ENABLE: + SDL_PauseAudio (0); + break; + + case VOICE_DISABLE: + SDL_PauseAudio (1); + break; + } + return 0; +} + +static void *sdl_audio_init (void) +{ + SDLAudioState *s = &glob_sdl; + conf.nb_samples = audio_get_conf_int (QC_SDL_SAMPLES, conf.nb_samples); + + if (SDL_InitSubSystem (SDL_INIT_AUDIO)) { + dolog ("SDL failed to initialize audio subsystem\nReason: %s\n", + errstr ()); + return NULL; + } + + s->mutex = SDL_CreateMutex (); + if (!s->mutex) { + dolog ("Failed to create SDL mutex\nReason: %s\n", errstr ()); + SDL_QuitSubSystem (SDL_INIT_AUDIO); + return NULL; + } + + s->sem = SDL_CreateSemaphore (0); + if (!s->sem) { + dolog ("Failed to create SDL semaphore\nReason: %s\n", errstr ()); + SDL_DestroyMutex (s->mutex); + SDL_QuitSubSystem (SDL_INIT_AUDIO); + return NULL; + } + + return s; +} + +static void sdl_audio_fini (void *opaque) +{ + SDLAudioState *s = opaque; + sdl_close (s); + SDL_DestroySemaphore (s->sem); + SDL_DestroyMutex (s->mutex); + SDL_QuitSubSystem (SDL_INIT_AUDIO); +} + +struct pcm_ops sdl_pcm_ops = { + sdl_hw_init, + sdl_hw_fini, + sdl_hw_run, + sdl_hw_write, + sdl_hw_ctl +}; + +struct audio_output_driver sdl_output_driver = { + "sdl", + sdl_audio_init, + sdl_audio_fini, + &sdl_pcm_ops, + 1, + 1, + sizeof (SDLVoice) +}; diff --git a/audio/sdlaudio.h b/audio/sdlaudio.h new file mode 100644 index 0000000000..380d0da2a2 --- /dev/null +++ b/audio/sdlaudio.h @@ -0,0 +1,34 @@ +/* + * QEMU SDL audio output driver header + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#ifndef QEMU_SDLAUDIO_H +#define QEMU_SDLAUDIO_H + +typedef struct SDLVoice { + struct HWVoice hw; +} SDLVoice; + +extern struct pcm_ops sdl_pcm_ops; +extern struct audio_output_driver sdl_output_driver; + +#endif /* sdlaudio.h */ diff --git a/audio/wavaudio.c b/audio/wavaudio.c new file mode 100644 index 0000000000..dee4a060dd --- /dev/null +++ b/audio/wavaudio.c @@ -0,0 +1,200 @@ +/* + * QEMU WAV audio output driver + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include "vl.h" + +#define AUDIO_CAP "wav" +#include "audio/audio.h" +#include "audio/wavaudio.h" + +static struct { + const char *wav_path; +} conf = { + .wav_path = "qemu.wav" +}; + +static void wav_hw_run (HWVoice *hw) +{ + WAVVoice *wav = (WAVVoice *) hw; + int rpos, live, decr, samples; + uint8_t *dst; + st_sample_t *src; + int64_t now = qemu_get_clock (vm_clock); + int64_t ticks = now - wav->old_ticks; + int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec; + wav->old_ticks = now; + + if (bytes > INT_MAX) + samples = INT_MAX >> hw->shift; + else + samples = bytes >> hw->shift; + + live = pcm_hw_get_live (hw); + if (live <= 0) + return; + + decr = audio_MIN (live, samples); + samples = decr; + rpos = hw->rpos; + while (samples) { + int left_till_end_samples = hw->samples - rpos; + int convert_samples = audio_MIN (samples, left_till_end_samples); + + src = advance (hw->mix_buf, rpos * sizeof (st_sample_t)); + dst = advance (wav->pcm_buf, rpos << hw->shift); + + hw->clip (dst, src, convert_samples); + qemu_put_buffer (wav->f, dst, convert_samples << hw->shift); + memset (src, 0, convert_samples * sizeof (st_sample_t)); + + rpos = (rpos + convert_samples) % hw->samples; + samples -= convert_samples; + wav->total_samples += convert_samples; + } + + pcm_hw_dec_live (hw, decr); + hw->rpos = rpos; +} + +static int wav_hw_write (SWVoice *sw, void *buf, int len) +{ + return pcm_hw_write (sw, buf, len); +} + + +/* VICE code: Store number as little endian. */ +static void le_store (uint8_t *buf, uint32_t val, int len) +{ + int i; + for (i = 0; i < len; i++) { + buf[i] = (uint8_t) (val & 0xff); + val >>= 8; + } +} + +static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt) +{ + WAVVoice *wav = (WAVVoice *) hw; + int bits16 = 0, stereo = audio_state.fixed_channels == 2; + uint8_t hdr[] = { + 0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, + 0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, + 0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04, + 0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00 + }; + + switch (audio_state.fixed_fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + break; + + case AUD_FMT_S16: + case AUD_FMT_U16: + bits16 = 1; + break; + } + + hdr[34] = bits16 ? 0x10 : 0x08; + hw->freq = 44100; + hw->nchannels = stereo ? 2 : 1; + hw->fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8; + hw->bufsize = 4096; + wav->pcm_buf = qemu_mallocz (hw->bufsize); + if (!wav->pcm_buf) + return -1; + + le_store (hdr + 22, hw->nchannels, 2); + le_store (hdr + 24, hw->freq, 4); + le_store (hdr + 28, hw->freq << (bits16 + stereo), 4); + le_store (hdr + 32, 1 << (bits16 + stereo), 2); + + wav->f = fopen (conf.wav_path, "wb"); + if (!wav->f) { + dolog ("failed to open wave file `%s'\nReason: %s\n", + conf.wav_path, strerror (errno)); + return -1; + } + + qemu_put_buffer (wav->f, hdr, sizeof (hdr)); + return 0; +} + +static void wav_hw_fini (HWVoice *hw) +{ + WAVVoice *wav = (WAVVoice *) hw; + int stereo = hw->nchannels == 2; + uint8_t rlen[4]; + uint8_t dlen[4]; + uint32_t rifflen = (wav->total_samples << stereo) + 36; + uint32_t datalen = wav->total_samples << stereo; + + if (!wav->f || !hw->active) + return; + + le_store (rlen, rifflen, 4); + le_store (dlen, datalen, 4); + + qemu_fseek (wav->f, 4, SEEK_SET); + qemu_put_buffer (wav->f, rlen, 4); + + qemu_fseek (wav->f, 32, SEEK_CUR); + qemu_put_buffer (wav->f, dlen, 4); + + fclose (wav->f); + wav->f = NULL; +} + +static int wav_hw_ctl (HWVoice *hw, int cmd, ...) +{ + (void) hw; + (void) cmd; + return 0; +} + +static void *wav_audio_init (void) +{ + return &conf; +} + +static void wav_audio_fini (void *opaque) +{ + ldebug ("wav_fini"); +} + +struct pcm_ops wav_pcm_ops = { + wav_hw_init, + wav_hw_fini, + wav_hw_run, + wav_hw_write, + wav_hw_ctl +}; + +struct audio_output_driver wav_output_driver = { + "wav", + wav_audio_init, + wav_audio_fini, + &wav_pcm_ops, + 1, + 1, + sizeof (WAVVoice) +}; diff --git a/audio/wavaudio.h b/audio/wavaudio.h new file mode 100644 index 0000000000..0b6070be76 --- /dev/null +++ b/audio/wavaudio.h @@ -0,0 +1,38 @@ +/* + * QEMU WAV audio output driver header + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#ifndef QEMU_WAVAUDIO_H +#define QEMU_WAVAUDIO_H + +typedef struct WAVVoice { + struct HWVoice hw; + QEMUFile *f; + int64_t old_ticks; + void *pcm_buf; + int total_samples; +} WAVVoice; + +extern struct pcm_ops wav_pcm_ops; +extern struct audio_output_driver wav_output_driver; + +#endif /* wavaudio.h */ diff --git a/hw/adlib.c b/hw/adlib.c new file mode 100644 index 0000000000..a49b32b53d --- /dev/null +++ b/hw/adlib.c @@ -0,0 +1,309 @@ +/* + * QEMU Adlib emulation + * + * Copyright (c) 2004 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include "vl.h" + +#define AUDIO_CAP "adlib" +#include "audio/audio.h" + +#ifdef USE_YMF262 +#define HAS_YMF262 1 +#include "ymf262.h" +void YMF262UpdateOneQEMU(int which, INT16 *dst, int length); +#define SHIFT 2 +#else +#include "fmopl.h" +#define SHIFT 1 +#endif + +#ifdef _WIN32 +#include <windows.h> +#define small_delay() Sleep (1) +#else +#define small_delay() usleep (1) +#endif + +#define IO_READ_PROTO(name) \ + uint32_t name (void *opaque, uint32_t nport) +#define IO_WRITE_PROTO(name) \ + void name (void *opaque, uint32_t nport, uint32_t val) + +static struct { + int port; + int freq; +} conf = {0x220, 44100}; + +typedef struct { + int enabled; + int active; + int cparam; + int64_t ticks; + int bufpos; + int16_t *mixbuf; + double interval; + QEMUTimer *ts, *opl_ts; + SWVoice *voice; + int left, pos, samples, bytes_per_second, old_free; + int refcount; +#ifndef USE_YMF262 + FM_OPL *opl; +#endif +} AdlibState; + +static AdlibState adlib; + +static IO_WRITE_PROTO(adlib_write) +{ + AdlibState *s = opaque; + int a = nport & 3; + int status; + + s->ticks = qemu_get_clock (vm_clock); + s->active = 1; + AUD_enable (s->voice, 1); + +#ifdef USE_YMF262 + status = YMF262Write (0, a, val); +#else + status = OPLWrite (s->opl, a, val); +#endif +} + +static IO_READ_PROTO(adlib_read) +{ + AdlibState *s = opaque; + uint8_t data; + int a = nport & 3; + +#ifdef USE_YMF262 + (void) s; + data = YMF262Read (0, a); +#else + data = OPLRead (s->opl, a); +#endif + return data; +} + +static void OPL_timer (void *opaque) +{ + AdlibState *s = opaque; +#ifdef USE_YMF262 + YMF262TimerOver (s->cparam >> 1, s->cparam & 1); +#else + OPLTimerOver (s->opl, s->cparam); +#endif + qemu_mod_timer (s->opl_ts, qemu_get_clock (vm_clock) + s->interval); +} + +static void YMF262TimerHandler (int c, double interval_Sec) +{ + AdlibState *s = &adlib; + if (interval_Sec == 0.0) { + qemu_del_timer (s->opl_ts); + return; + } + s->cparam = c; + s->interval = ticks_per_sec * interval_Sec; + qemu_mod_timer (s->opl_ts, qemu_get_clock (vm_clock) + s->interval); + small_delay (); +} + +static int write_audio (AdlibState *s, int samples) +{ + int net = 0; + int ss = samples; + while (samples) { + int nbytes = samples << SHIFT; + int wbytes = AUD_write (s->voice, + s->mixbuf + (s->pos << (SHIFT - 1)), + nbytes); + int wsampl = wbytes >> SHIFT; + samples -= wsampl; + s->pos = (s->pos + wsampl) % s->samples; + net += wsampl; + if (!wbytes) + break; + } + if (net > ss) { + dolog ("WARNING: net > ss\n"); + } + return net; +} + +static void timer (void *opaque) +{ + AdlibState *s = opaque; + int elapsed, samples, net = 0; + + if (s->refcount) + dolog ("refcount=%d\n", s->refcount); + + s->refcount += 1; + if (!(s->active && s->enabled)) + goto reset; + + AUD_run (); + + while (s->left) { + int written = write_audio (s, s->left); + net += written; + if (!written) + goto reset2; + s->left -= written; + } + s->pos = 0; + + elapsed = AUD_calc_elapsed (s->voice); + if (!elapsed) + goto reset2; + + /* elapsed = AUD_get_free (s->voice); */ + samples = elapsed >> SHIFT; + if (!samples) + goto reset2; + + samples = audio_MIN (samples, s->samples - s->pos); + if (s->left) + dolog ("left=%d samples=%d elapsed=%d free=%d\n", + s->left, samples, elapsed, AUD_get_free (s->voice)); + + if (!samples) + goto reset2; + +#ifdef USE_YMF262 + YMF262UpdateOneQEMU (0, s->mixbuf + s->pos * 2, samples); +#else + YM3812UpdateOne (s->opl, s->mixbuf + s->pos, samples); +#endif + + while (samples) { + int written = write_audio (s, samples); + net += written; + if (!written) + break; + samples -= written; + } + if (!samples) + s->pos = 0; + s->left = samples; + +reset2: + AUD_adjust (s->voice, net << SHIFT); +reset: + qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + ticks_per_sec / 1024); + s->refcount -= 1; +} + +static void Adlib_fini (AdlibState *s) +{ +#ifdef USE_YMF262 + YMF262Shutdown (); +#else + if (s->opl) { + OPLDestroy (s->opl); + s->opl = NULL; + } +#endif + + if (s->opl_ts) + qemu_free_timer (s->opl_ts); + + if (s->ts) + qemu_free_timer (s->ts); + +#define maybe_free(p) if (p) qemu_free (p) + maybe_free (s->mixbuf); +#undef maybe_free + + s->active = 0; + s->enabled = 0; +} + +void Adlib_init (void) +{ + AdlibState *s = &adlib; + + memset (s, 0, sizeof (*s)); + +#ifdef USE_YMF262 + if (YMF262Init (1, 14318180, conf.freq)) { + dolog ("YMF262Init %d failed\n", conf.freq); + return; + } + else { + YMF262SetTimerHandler (0, YMF262TimerHandler, 0); + s->enabled = 1; + } +#else + s->opl = OPLCreate (OPL_TYPE_YM3812, 3579545, conf.freq); + if (!s->opl) { + dolog ("OPLCreate %d failed\n", conf.freq); + return; + } + else { + OPLSetTimerHandler (s->opl, YMF262TimerHandler, 0); + s->enabled = 1; + } +#endif + + s->opl_ts = qemu_new_timer (vm_clock, OPL_timer, s); + if (!s->opl_ts) { + dolog ("Can not get timer for adlib emulation\n"); + Adlib_fini (s); + return; + } + + s->ts = qemu_new_timer (vm_clock, timer, s); + if (!s->opl_ts) { + dolog ("Can not get timer for adlib emulation\n"); + Adlib_fini (s); + return; + } + + s->voice = AUD_open (s->voice, "adlib", conf.freq, SHIFT, AUD_FMT_S16); + if (!s->voice) { + Adlib_fini (s); + return; + } + + s->bytes_per_second = conf.freq << SHIFT; + s->samples = AUD_get_buffer_size (s->voice) >> SHIFT; + s->mixbuf = qemu_mallocz (s->samples << SHIFT); + + if (!s->mixbuf) { + dolog ("not enough memory for adlib mixing buffer (%d)\n", + s->samples << SHIFT); + Adlib_fini (s); + return; + } + register_ioport_read (0x388, 4, 1, adlib_read, s); + register_ioport_write (0x388, 4, 1, adlib_write, s); + + register_ioport_read (conf.port, 4, 1, adlib_read, s); + register_ioport_write (conf.port, 4, 1, adlib_write, s); + + register_ioport_read (conf.port + 8, 2, 1, adlib_read, s); + register_ioport_write (conf.port + 8, 2, 1, adlib_write, s); + + qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1); +} @@ -1,8 +1,8 @@ /* * QEMU DMA emulation - * - * Copyright (c) 2003 Vassili Karpov (malc) - * + * + * Copyright (c) 2003-2004 Vassili Karpov (malc) + * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights @@ -23,9 +23,9 @@ */ #include "vl.h" -//#define DEBUG_DMA +/* #define DEBUG_DMA */ -#define log(...) fprintf (stderr, "dma: " __VA_ARGS__) +#define dolog(...) fprintf (stderr, "dma: " __VA_ARGS__) #ifdef DEBUG_DMA #define lwarn(...) fprintf (stderr, "dma: " __VA_ARGS__) #define linfo(...) fprintf (stderr, "dma: " __VA_ARGS__) @@ -86,7 +86,7 @@ static void write_page (void *opaque, uint32_t nport, uint32_t data) ichan = channels[nport & 7]; if (-1 == ichan) { - log ("invalid channel %#x %#x\n", nport, data); + dolog ("invalid channel %#x %#x\n", nport, data); return; } d->regs[ichan].page = data; @@ -99,7 +99,7 @@ static void write_pageh (void *opaque, uint32_t nport, uint32_t data) ichan = channels[nport & 7]; if (-1 == ichan) { - log ("invalid channel %#x %#x\n", nport, data); + dolog ("invalid channel %#x %#x\n", nport, data); return; } d->regs[ichan].pageh = data; @@ -112,7 +112,7 @@ static uint32_t read_page (void *opaque, uint32_t nport) ichan = channels[nport & 7]; if (-1 == ichan) { - log ("invalid channel read %#x\n", nport); + dolog ("invalid channel read %#x\n", nport); return 0; } return d->regs[ichan].page; @@ -125,7 +125,7 @@ static uint32_t read_pageh (void *opaque, uint32_t nport) ichan = channels[nport & 7]; if (-1 == ichan) { - log ("invalid channel read %#x\n", nport); + dolog ("invalid channel read %#x\n", nport); return 0; } return d->regs[ichan].pageh; @@ -136,7 +136,7 @@ static inline void init_chan (struct dma_cont *d, int ichan) struct dma_regs *r; r = d->regs + ichan; - r->now[ADDR] = r->base[0] << d->dshift; + r->now[ADDR] = r->base[ADDR] << d->dshift; r->now[COUNT] = 0; } @@ -152,7 +152,7 @@ static inline int getff (struct dma_cont *d) static uint32_t read_chan (void *opaque, uint32_t nport) { struct dma_cont *d = opaque; - int ichan, nreg, iport, ff, val; + int ichan, nreg, iport, ff, val, dir; struct dma_regs *r; iport = (nport >> d->dshift) & 0x0f; @@ -160,12 +160,14 @@ static uint32_t read_chan (void *opaque, uint32_t nport) nreg = iport & 1; r = d->regs + ichan; + dir = ((r->mode >> 5) & 1) ? -1 : 1; ff = getff (d); if (nreg) val = (r->base[COUNT] << d->dshift) - r->now[COUNT]; else - val = r->now[ADDR] + r->now[COUNT]; + val = r->now[ADDR] + r->now[COUNT] * dir; + ldebug ("read_chan %#x -> %d\n", iport, val); return (val >> (d->dshift + (ff << 3))) & 0xff; } @@ -190,19 +192,19 @@ static void write_chan (void *opaque, uint32_t nport, uint32_t data) static void write_cont (void *opaque, uint32_t nport, uint32_t data) { struct dma_cont *d = opaque; - int iport, ichan; + int iport, ichan = 0; iport = (nport >> d->dshift) & 0x0f; switch (iport) { - case 8: /* command */ + case 0x08: /* command */ if ((data != 0) && (data & CMD_NOT_SUPPORTED)) { - log ("command %#x not supported\n", data); + dolog ("command %#x not supported\n", data); return; } d->command = data; break; - case 9: + case 0x09: ichan = data & 3; if (data & 4) { d->status |= 1 << (ichan + 4); @@ -213,22 +215,19 @@ static void write_cont (void *opaque, uint32_t nport, uint32_t data) d->status &= ~(1 << ichan); break; - case 0xa: /* single mask */ + case 0x0a: /* single mask */ if (data & 4) d->mask |= 1 << (data & 3); else d->mask &= ~(1 << (data & 3)); break; - case 0xb: /* mode */ + case 0x0b: /* mode */ { ichan = data & 3; #ifdef DEBUG_DMA - int op; - int ai; - int dir; - int opmode; - + { + int op, ai, dir, opmode; op = (data >> 2) & 3; ai = (data >> 4) & 1; dir = (data >> 5) & 1; @@ -236,39 +235,39 @@ static void write_cont (void *opaque, uint32_t nport, uint32_t data) linfo ("ichan %d, op %d, ai %d, dir %d, opmode %d\n", ichan, op, ai, dir, opmode); + } #endif - d->regs[ichan].mode = data; break; } - case 0xc: /* clear flip flop */ + case 0x0c: /* clear flip flop */ d->flip_flop = 0; break; - case 0xd: /* reset */ + case 0x0d: /* reset */ d->flip_flop = 0; d->mask = ~0; d->status = 0; d->command = 0; break; - case 0xe: /* clear mask for all channels */ + case 0x0e: /* clear mask for all channels */ d->mask = 0; break; - case 0xf: /* write mask for all channels */ + case 0x0f: /* write mask for all channels */ d->mask = data; break; default: - log ("dma: unknown iport %#x\n", iport); + dolog ("unknown iport %#x\n", iport); break; } #ifdef DEBUG_DMA if (0xc != iport) { - linfo ("nport %#06x, ichan % 2d, val %#06x\n", + linfo ("write_cont: nport %#06x, ichan % 2d, val %#06x\n", nport, ichan, data); } #endif @@ -278,20 +277,22 @@ static uint32_t read_cont (void *opaque, uint32_t nport) { struct dma_cont *d = opaque; int iport, val; - + iport = (nport >> d->dshift) & 0x0f; switch (iport) { - case 0x08: /* status */ + case 0x08: /* status */ val = d->status; d->status &= 0xf0; break; - case 0x0f: /* mask */ + case 0x0f: /* mask */ val = d->mask; break; default: val = 0; break; } + + ldebug ("read_cont: nport %#06x, iport %#04x val %#x\n", nport, iport, val); return val; } @@ -322,23 +323,27 @@ void DMA_release_DREQ (int nchan) static void channel_run (int ncont, int ichan) { - struct dma_regs *r; int n; - target_ulong addr; -/* int ai, dir; */ + struct dma_regs *r = &dma_controllers[ncont].regs[ichan]; +#ifdef DEBUG_DMA + int dir, opmode; - r = dma_controllers[ncont].regs + ichan; -/* ai = r->mode & 16; */ -/* dir = r->mode & 32 ? -1 : 1; */ + dir = (r->mode >> 5) & 1; + opmode = (r->mode >> 6) & 3; - /* NOTE: pageh is only used by PPC PREP */ - addr = ((r->pageh & 0x7f) << 24) | (r->page << 16) | r->now[ADDR]; - n = r->transfer_handler (r->opaque, addr, - (r->base[COUNT] << ncont) + (1 << ncont)); - r->now[COUNT] = n; + if (dir) { + dolog ("DMA in address decrement mode\n"); + } + if (opmode != 1) { + dolog ("DMA not in single mode select %#x\n", opmode); + } +#endif - ldebug ("dma_pos %d size %d\n", - n, (r->base[1] << ncont) + (1 << ncont)); + r = dma_controllers[ncont].regs + ichan; + n = r->transfer_handler (r->opaque, ichan + (ncont << 2), + r->now[COUNT], (r->base[COUNT] + 1) << ncont); + r->now[COUNT] = n; + ldebug ("dma_pos %d size %d\n", n, (r->base[COUNT] + 1) << ncont); } void DMA_run (void) @@ -361,7 +366,7 @@ void DMA_run (void) } void DMA_register_channel (int nchan, - DMA_transfer_handler transfer_handler, + DMA_transfer_handler transfer_handler, void *opaque) { struct dma_regs *r; @@ -375,6 +380,50 @@ void DMA_register_channel (int nchan, r->opaque = opaque; } +int DMA_read_memory (int nchan, void *buf, int pos, int len) +{ + struct dma_regs *r = &dma_controllers[nchan > 3].regs[nchan & 3]; + target_ulong addr = ((r->pageh & 0x7f) << 24) | (r->page << 16) | r->now[ADDR]; + + if (r->mode & 0x20) { + int i; + uint8_t *p = buf; + + cpu_physical_memory_read (addr - pos - len, buf, len); + /* What about 16bit transfers? */ + for (i = 0; i < len >> 1; i++) { + uint8_t b = p[len - i - 1]; + p[i] = b; + } + } + else + cpu_physical_memory_read (addr + pos, buf, len); + + return len; +} + +int DMA_write_memory (int nchan, void *buf, int pos, int len) +{ + struct dma_regs *r = &dma_controllers[nchan > 3].regs[nchan & 3]; + target_ulong addr = ((r->pageh & 0x7f) << 24) | (r->page << 16) | r->now[ADDR]; + + if (r->mode & 0x20) { + int i; + uint8_t *p = buf; + + cpu_physical_memory_write (addr - pos - len, buf, len); + /* What about 16bit transfers? */ + for (i = 0; i < len; i++) { + uint8_t b = p[len - i - 1]; + p[i] = b; + } + } + else + cpu_physical_memory_write (addr + pos, buf, len); + + return len; +} + /* request the emulator to transfer a new DMA memory block ASAP */ void DMA_schedule(int nchan) { @@ -388,7 +437,7 @@ static void dma_reset(void *opaque) } /* dshift = 0: 8 bit DMA, 1 = 16 bit DMA */ -static void dma_init2(struct dma_cont *d, int base, int dshift, +static void dma_init2(struct dma_cont *d, int base, int dshift, int page_base, int pageh_base) { const static int page_port_list[] = { 0x1, 0x2, 0x3, 0x7 }; @@ -400,31 +449,87 @@ static void dma_init2(struct dma_cont *d, int base, int dshift, register_ioport_read (base + (i << dshift), 1, 1, read_chan, d); } for (i = 0; i < LENOFA (page_port_list); i++) { - register_ioport_write (page_base + page_port_list[i], 1, 1, + register_ioport_write (page_base + page_port_list[i], 1, 1, write_page, d); - register_ioport_read (page_base + page_port_list[i], 1, 1, + register_ioport_read (page_base + page_port_list[i], 1, 1, read_page, d); if (pageh_base >= 0) { - register_ioport_write (pageh_base + page_port_list[i], 1, 1, + register_ioport_write (pageh_base + page_port_list[i], 1, 1, write_pageh, d); - register_ioport_read (pageh_base + page_port_list[i], 1, 1, + register_ioport_read (pageh_base + page_port_list[i], 1, 1, read_pageh, d); } } for (i = 0; i < 8; i++) { - register_ioport_write (base + ((i + 8) << dshift), 1, 1, + register_ioport_write (base + ((i + 8) << dshift), 1, 1, write_cont, d); - register_ioport_read (base + ((i + 8) << dshift), 1, 1, + register_ioport_read (base + ((i + 8) << dshift), 1, 1, read_cont, d); } qemu_register_reset(dma_reset, d); dma_reset(d); } +static void dma_save (QEMUFile *f, void *opaque) +{ + struct dma_cont *d = opaque; + int i; + + /* qemu_put_8s (f, &d->status); */ + qemu_put_8s (f, &d->command); + qemu_put_8s (f, &d->mask); + qemu_put_8s (f, &d->flip_flop); + qemu_put_be32s (f, &d->dshift); + + for (i = 0; i < 4; ++i) { + struct dma_regs *r = &d->regs[i]; + qemu_put_be32s (f, &r->now[0]); + qemu_put_be32s (f, &r->now[1]); + qemu_put_be16s (f, &r->base[0]); + qemu_put_be16s (f, &r->base[1]); + qemu_put_8s (f, &r->mode); + qemu_put_8s (f, &r->page); + qemu_put_8s (f, &r->pageh); + qemu_put_8s (f, &r->dack); + qemu_put_8s (f, &r->eop); + } +} + +static int dma_load (QEMUFile *f, void *opaque, int version_id) +{ + struct dma_cont *d = opaque; + int i; + + if (version_id != 1) + return -EINVAL; + + /* qemu_get_8s (f, &d->status); */ + qemu_get_8s (f, &d->command); + qemu_get_8s (f, &d->mask); + qemu_get_8s (f, &d->flip_flop); + qemu_get_be32s (f, &d->dshift); + + for (i = 0; i < 4; ++i) { + struct dma_regs *r = &d->regs[i]; + qemu_get_be32s (f, &r->now[0]); + qemu_get_be32s (f, &r->now[1]); + qemu_get_be16s (f, &r->base[0]); + qemu_get_be16s (f, &r->base[1]); + qemu_get_8s (f, &r->mode); + qemu_get_8s (f, &r->page); + qemu_get_8s (f, &r->pageh); + qemu_get_8s (f, &r->dack); + qemu_get_8s (f, &r->eop); + } + return 0; +} + void DMA_init (int high_page_enable) { - dma_init2(&dma_controllers[0], 0x00, 0, 0x80, + dma_init2(&dma_controllers[0], 0x00, 0, 0x80, high_page_enable ? 0x480 : -1); dma_init2(&dma_controllers[1], 0xc0, 1, 0x88, high_page_enable ? 0x488 : -1); + register_savevm ("dma", 0, 1, dma_save, dma_load, &dma_controllers[0]); + register_savevm ("dma", 1, 1, dma_save, dma_load, &dma_controllers[1]); } @@ -313,7 +313,8 @@ static void fd_reset (fdrive_t *drv) static void fdctrl_reset (fdctrl_t *fdctrl, int do_irq); static void fdctrl_reset_fifo (fdctrl_t *fdctrl); -static int fdctrl_transfer_handler (void *opaque, target_ulong addr, int size); +static int fdctrl_transfer_handler (void *opaque, int nchan, + int dma_pos, int dma_len); static void fdctrl_raise_irq (fdctrl_t *fdctrl, uint8_t status); static void fdctrl_result_timer(void *opaque); @@ -908,7 +909,8 @@ static void fdctrl_start_transfer_del (fdctrl_t *fdctrl, int direction) } /* handlers for DMA transfers */ -static int fdctrl_transfer_handler (void *opaque, target_ulong addr, int size) +static int fdctrl_transfer_handler (void *opaque, int nchan, + int dma_pos, int dma_len) { fdctrl_t *fdctrl; fdrive_t *cur_drv; @@ -924,8 +926,8 @@ static int fdctrl_transfer_handler (void *opaque, target_ulong addr, int size) if (fdctrl->data_dir == FD_DIR_SCANE || fdctrl->data_dir == FD_DIR_SCANL || fdctrl->data_dir == FD_DIR_SCANH) status2 = 0x04; - if (size > fdctrl->data_len) - size = fdctrl->data_len; + if (dma_len > fdctrl->data_len) + dma_len = fdctrl->data_len; if (cur_drv->bs == NULL) { if (fdctrl->data_dir == FD_DIR_WRITE) fdctrl_stop_transfer(fdctrl, 0x60, 0x00, 0x00); @@ -935,8 +937,8 @@ static int fdctrl_transfer_handler (void *opaque, target_ulong addr, int size) goto transfer_error; } rel_pos = fdctrl->data_pos % FD_SECTOR_LEN; - for (start_pos = fdctrl->data_pos; fdctrl->data_pos < size;) { - len = size - fdctrl->data_pos; + for (start_pos = fdctrl->data_pos; fdctrl->data_pos < dma_len;) { + len = dma_len - fdctrl->data_pos; if (len + rel_pos > FD_SECTOR_LEN) len = FD_SECTOR_LEN - rel_pos; FLOPPY_DPRINTF("copy %d bytes (%d %d %d) %d pos %d %02x %02x " @@ -958,13 +960,17 @@ static int fdctrl_transfer_handler (void *opaque, target_ulong addr, int size) switch (fdctrl->data_dir) { case FD_DIR_READ: /* READ commands */ - cpu_physical_memory_write(addr + fdctrl->data_pos, - fdctrl->fifo + rel_pos, len); + DMA_write_memory (nchan, fdctrl->fifo + rel_pos, + fdctrl->data_pos, len); +/* cpu_physical_memory_write(addr + fdctrl->data_pos, */ +/* fdctrl->fifo + rel_pos, len); */ break; case FD_DIR_WRITE: /* WRITE commands */ - cpu_physical_memory_read(addr + fdctrl->data_pos, - fdctrl->fifo + rel_pos, len); + DMA_read_memory (nchan, fdctrl->fifo + rel_pos, + fdctrl->data_pos, len); +/* cpu_physical_memory_read(addr + fdctrl->data_pos, */ +/* fdctrl->fifo + rel_pos, len); */ if (bdrv_write(cur_drv->bs, fd_sector(cur_drv), fdctrl->fifo, 1) < 0) { FLOPPY_ERROR("writting sector %d\n", fd_sector(cur_drv)); @@ -977,8 +983,9 @@ static int fdctrl_transfer_handler (void *opaque, target_ulong addr, int size) { uint8_t tmpbuf[FD_SECTOR_LEN]; int ret; - cpu_physical_memory_read(addr + fdctrl->data_pos, - tmpbuf, len); + DMA_read_memory (nchan, tmpbuf, fdctrl->data_pos, len); +/* cpu_physical_memory_read(addr + fdctrl->data_pos, */ +/* tmpbuf, len); */ ret = memcmp(tmpbuf, fdctrl->fifo + rel_pos, len); if (ret == 0) { status2 = 0x08; diff --git a/hw/fmopl.c b/hw/fmopl.c new file mode 100644 index 0000000000..2b0e82b0cc --- /dev/null +++ b/hw/fmopl.c @@ -0,0 +1,1390 @@ +/* +** +** File: fmopl.c -- software implementation of FM sound generator +** +** Copyright (C) 1999,2000 Tatsuyuki Satoh , MultiArcadeMachineEmurator development +** +** Version 0.37a +** +*/ + +/* + preliminary : + Problem : + note: +*/ + +/* This version of fmopl.c is a fork of the MAME one, relicensed under the LGPL. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#define INLINE __inline +#define HAS_YM3812 1 + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <stdarg.h> +#include <math.h> +//#include "driver.h" /* use M.A.M.E. */ +#include "fmopl.h" + +#ifndef PI +#define PI 3.14159265358979323846 +#endif + +/* -------------------- for debug --------------------- */ +/* #define OPL_OUTPUT_LOG */ +#ifdef OPL_OUTPUT_LOG +static FILE *opl_dbg_fp = NULL; +static FM_OPL *opl_dbg_opl[16]; +static int opl_dbg_maxchip,opl_dbg_chip; +#endif + +/* -------------------- preliminary define section --------------------- */ +/* attack/decay rate time rate */ +#define OPL_ARRATE 141280 /* RATE 4 = 2826.24ms @ 3.6MHz */ +#define OPL_DRRATE 1956000 /* RATE 4 = 39280.64ms @ 3.6MHz */ + +#define DELTAT_MIXING_LEVEL (1) /* DELTA-T ADPCM MIXING LEVEL */ + +#define FREQ_BITS 24 /* frequency turn */ + +/* counter bits = 20 , octerve 7 */ +#define FREQ_RATE (1<<(FREQ_BITS-20)) +#define TL_BITS (FREQ_BITS+2) + +/* final output shift , limit minimum and maximum */ +#define OPL_OUTSB (TL_BITS+3-16) /* OPL output final shift 16bit */ +#define OPL_MAXOUT (0x7fff<<OPL_OUTSB) +#define OPL_MINOUT (-0x8000<<OPL_OUTSB) + +/* -------------------- quality selection --------------------- */ + +/* sinwave entries */ +/* used static memory = SIN_ENT * 4 (byte) */ +#define SIN_ENT 2048 + +/* output level entries (envelope,sinwave) */ +/* envelope counter lower bits */ +#define ENV_BITS 16 +/* envelope output entries */ +#define EG_ENT 4096 +/* used dynamic memory = EG_ENT*4*4(byte)or EG_ENT*6*4(byte) */ +/* used static memory = EG_ENT*4 (byte) */ + +#define EG_OFF ((2*EG_ENT)<<ENV_BITS) /* OFF */ +#define EG_DED EG_OFF +#define EG_DST (EG_ENT<<ENV_BITS) /* DECAY START */ +#define EG_AED EG_DST +#define EG_AST 0 /* ATTACK START */ + +#define EG_STEP (96.0/EG_ENT) /* OPL is 0.1875 dB step */ + +/* LFO table entries */ +#define VIB_ENT 512 +#define VIB_SHIFT (32-9) +#define AMS_ENT 512 +#define AMS_SHIFT (32-9) + +#define VIB_RATE 256 + +/* -------------------- local defines , macros --------------------- */ + +/* register number to channel number , slot offset */ +#define SLOT1 0 +#define SLOT2 1 + +/* envelope phase */ +#define ENV_MOD_RR 0x00 +#define ENV_MOD_DR 0x01 +#define ENV_MOD_AR 0x02 + +/* -------------------- tables --------------------- */ +static const int slot_array[32]= +{ + 0, 2, 4, 1, 3, 5,-1,-1, + 6, 8,10, 7, 9,11,-1,-1, + 12,14,16,13,15,17,-1,-1, + -1,-1,-1,-1,-1,-1,-1,-1 +}; + +/* key scale level */ +/* table is 3dB/OCT , DV converts this in TL step at 6dB/OCT */ +#define DV (EG_STEP/2) +static const UINT32 KSL_TABLE[8*16]= +{ + /* OCT 0 */ + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + /* OCT 1 */ + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + 0.000/DV, 0.750/DV, 1.125/DV, 1.500/DV, + 1.875/DV, 2.250/DV, 2.625/DV, 3.000/DV, + /* OCT 2 */ + 0.000/DV, 0.000/DV, 0.000/DV, 0.000/DV, + 0.000/DV, 1.125/DV, 1.875/DV, 2.625/DV, + 3.000/DV, 3.750/DV, 4.125/DV, 4.500/DV, + 4.875/DV, 5.250/DV, 5.625/DV, 6.000/DV, + /* OCT 3 */ + 0.000/DV, 0.000/DV, 0.000/DV, 1.875/DV, + 3.000/DV, 4.125/DV, 4.875/DV, 5.625/DV, + 6.000/DV, 6.750/DV, 7.125/DV, 7.500/DV, + 7.875/DV, 8.250/DV, 8.625/DV, 9.000/DV, + /* OCT 4 */ + 0.000/DV, 0.000/DV, 3.000/DV, 4.875/DV, + 6.000/DV, 7.125/DV, 7.875/DV, 8.625/DV, + 9.000/DV, 9.750/DV,10.125/DV,10.500/DV, + 10.875/DV,11.250/DV,11.625/DV,12.000/DV, + /* OCT 5 */ + 0.000/DV, 3.000/DV, 6.000/DV, 7.875/DV, + 9.000/DV,10.125/DV,10.875/DV,11.625/DV, + 12.000/DV,12.750/DV,13.125/DV,13.500/DV, + 13.875/DV,14.250/DV,14.625/DV,15.000/DV, + /* OCT 6 */ + 0.000/DV, 6.000/DV, 9.000/DV,10.875/DV, + 12.000/DV,13.125/DV,13.875/DV,14.625/DV, + 15.000/DV,15.750/DV,16.125/DV,16.500/DV, + 16.875/DV,17.250/DV,17.625/DV,18.000/DV, + /* OCT 7 */ + 0.000/DV, 9.000/DV,12.000/DV,13.875/DV, + 15.000/DV,16.125/DV,16.875/DV,17.625/DV, + 18.000/DV,18.750/DV,19.125/DV,19.500/DV, + 19.875/DV,20.250/DV,20.625/DV,21.000/DV +}; +#undef DV + +/* sustain lebel table (3db per step) */ +/* 0 - 15: 0, 3, 6, 9,12,15,18,21,24,27,30,33,36,39,42,93 (dB)*/ +#define SC(db) (db*((3/EG_STEP)*(1<<ENV_BITS)))+EG_DST +static const INT32 SL_TABLE[16]={ + SC( 0),SC( 1),SC( 2),SC(3 ),SC(4 ),SC(5 ),SC(6 ),SC( 7), + SC( 8),SC( 9),SC(10),SC(11),SC(12),SC(13),SC(14),SC(31) +}; +#undef SC + +#define TL_MAX (EG_ENT*2) /* limit(tl + ksr + envelope) + sinwave */ +/* TotalLevel : 48 24 12 6 3 1.5 0.75 (dB) */ +/* TL_TABLE[ 0 to TL_MAX ] : plus section */ +/* TL_TABLE[ TL_MAX to TL_MAX+TL_MAX-1 ] : minus section */ +static INT32 *TL_TABLE; + +/* pointers to TL_TABLE with sinwave output offset */ +static INT32 **SIN_TABLE; + +/* LFO table */ +static INT32 *AMS_TABLE; +static INT32 *VIB_TABLE; + +/* envelope output curve table */ +/* attack + decay + OFF */ +static INT32 ENV_CURVE[2*EG_ENT+1]; + +/* multiple table */ +#define ML 2 +static const UINT32 MUL_TABLE[16]= { +/* 1/2, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15 */ + 0.50*ML, 1.00*ML, 2.00*ML, 3.00*ML, 4.00*ML, 5.00*ML, 6.00*ML, 7.00*ML, + 8.00*ML, 9.00*ML,10.00*ML,10.00*ML,12.00*ML,12.00*ML,15.00*ML,15.00*ML +}; +#undef ML + +/* dummy attack / decay rate ( when rate == 0 ) */ +static INT32 RATE_0[16]= +{0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; + +/* -------------------- static state --------------------- */ + +/* lock level of common table */ +static int num_lock = 0; + +/* work table */ +static void *cur_chip = NULL; /* current chip point */ +/* currenct chip state */ +/* static OPLSAMPLE *bufL,*bufR; */ +static OPL_CH *S_CH; +static OPL_CH *E_CH; +OPL_SLOT *SLOT7_1,*SLOT7_2,*SLOT8_1,*SLOT8_2; + +static INT32 outd[1]; +static INT32 ams; +static INT32 vib; +INT32 *ams_table; +INT32 *vib_table; +static INT32 amsIncr; +static INT32 vibIncr; +static INT32 feedback2; /* connect for SLOT 2 */ + +/* log output level */ +#define LOG_ERR 3 /* ERROR */ +#define LOG_WAR 2 /* WARNING */ +#define LOG_INF 1 /* INFORMATION */ + +//#define LOG_LEVEL LOG_INF +#define LOG_LEVEL LOG_ERR + +//#define LOG(n,x) if( (n)>=LOG_LEVEL ) logerror x +#define LOG(n,x) + +/* --------------------- subroutines --------------------- */ + +INLINE int Limit( int val, int max, int min ) { + if ( val > max ) + val = max; + else if ( val < min ) + val = min; + + return val; +} + +/* status set and IRQ handling */ +INLINE void OPL_STATUS_SET(FM_OPL *OPL,int flag) +{ + /* set status flag */ + OPL->status |= flag; + if(!(OPL->status & 0x80)) + { + if(OPL->status & OPL->statusmask) + { /* IRQ on */ + OPL->status |= 0x80; + /* callback user interrupt handler (IRQ is OFF to ON) */ + if(OPL->IRQHandler) (OPL->IRQHandler)(OPL->IRQParam,1); + } + } +} + +/* status reset and IRQ handling */ +INLINE void OPL_STATUS_RESET(FM_OPL *OPL,int flag) +{ + /* reset status flag */ + OPL->status &=~flag; + if((OPL->status & 0x80)) + { + if (!(OPL->status & OPL->statusmask) ) + { + OPL->status &= 0x7f; + /* callback user interrupt handler (IRQ is ON to OFF) */ + if(OPL->IRQHandler) (OPL->IRQHandler)(OPL->IRQParam,0); + } + } +} + +/* IRQ mask set */ +INLINE void OPL_STATUSMASK_SET(FM_OPL *OPL,int flag) +{ + OPL->statusmask = flag; + /* IRQ handling check */ + OPL_STATUS_SET(OPL,0); + OPL_STATUS_RESET(OPL,0); +} + +/* ----- key on ----- */ +INLINE void OPL_KEYON(OPL_SLOT *SLOT) +{ + /* sin wave restart */ + SLOT->Cnt = 0; + /* set attack */ + SLOT->evm = ENV_MOD_AR; + SLOT->evs = SLOT->evsa; + SLOT->evc = EG_AST; + SLOT->eve = EG_AED; +} +/* ----- key off ----- */ +INLINE void OPL_KEYOFF(OPL_SLOT *SLOT) +{ + if( SLOT->evm > ENV_MOD_RR) + { + /* set envelope counter from envleope output */ + SLOT->evm = ENV_MOD_RR; + if( !(SLOT->evc&EG_DST) ) + //SLOT->evc = (ENV_CURVE[SLOT->evc>>ENV_BITS]<<ENV_BITS) + EG_DST; + SLOT->evc = EG_DST; + SLOT->eve = EG_DED; + SLOT->evs = SLOT->evsr; + } +} + +/* ---------- calcrate Envelope Generator & Phase Generator ---------- */ +/* return : envelope output */ +INLINE UINT32 OPL_CALC_SLOT( OPL_SLOT *SLOT ) +{ + /* calcrate envelope generator */ + if( (SLOT->evc+=SLOT->evs) >= SLOT->eve ) + { + switch( SLOT->evm ){ + case ENV_MOD_AR: /* ATTACK -> DECAY1 */ + /* next DR */ + SLOT->evm = ENV_MOD_DR; + SLOT->evc = EG_DST; + SLOT->eve = SLOT->SL; + SLOT->evs = SLOT->evsd; + break; + case ENV_MOD_DR: /* DECAY -> SL or RR */ + SLOT->evc = SLOT->SL; + SLOT->eve = EG_DED; + if(SLOT->eg_typ) + { + SLOT->evs = 0; + } + else + { + SLOT->evm = ENV_MOD_RR; + SLOT->evs = SLOT->evsr; + } + break; + case ENV_MOD_RR: /* RR -> OFF */ + SLOT->evc = EG_OFF; + SLOT->eve = EG_OFF+1; + SLOT->evs = 0; + break; + } + } + /* calcrate envelope */ + return SLOT->TLL+ENV_CURVE[SLOT->evc>>ENV_BITS]+(SLOT->ams ? ams : 0); +} + +/* set algorythm connection */ +static void set_algorythm( OPL_CH *CH) +{ + INT32 *carrier = &outd[0]; + CH->connect1 = CH->CON ? carrier : &feedback2; + CH->connect2 = carrier; +} + +/* ---------- frequency counter for operater update ---------- */ +INLINE void CALC_FCSLOT(OPL_CH *CH,OPL_SLOT *SLOT) +{ + int ksr; + + /* frequency step counter */ + SLOT->Incr = CH->fc * SLOT->mul; + ksr = CH->kcode >> SLOT->KSR; + + if( SLOT->ksr != ksr ) + { + SLOT->ksr = ksr; + /* attack , decay rate recalcration */ + SLOT->evsa = SLOT->AR[ksr]; + SLOT->evsd = SLOT->DR[ksr]; + SLOT->evsr = SLOT->RR[ksr]; + } + SLOT->TLL = SLOT->TL + (CH->ksl_base>>SLOT->ksl); +} + +/* set multi,am,vib,EG-TYP,KSR,mul */ +INLINE void set_mul(FM_OPL *OPL,int slot,int v) +{ + OPL_CH *CH = &OPL->P_CH[slot/2]; + OPL_SLOT *SLOT = &CH->SLOT[slot&1]; + + SLOT->mul = MUL_TABLE[v&0x0f]; + SLOT->KSR = (v&0x10) ? 0 : 2; + SLOT->eg_typ = (v&0x20)>>5; + SLOT->vib = (v&0x40); + SLOT->ams = (v&0x80); + CALC_FCSLOT(CH,SLOT); +} + +/* set ksl & tl */ +INLINE void set_ksl_tl(FM_OPL *OPL,int slot,int v) +{ + OPL_CH *CH = &OPL->P_CH[slot/2]; + OPL_SLOT *SLOT = &CH->SLOT[slot&1]; + int ksl = v>>6; /* 0 / 1.5 / 3 / 6 db/OCT */ + + SLOT->ksl = ksl ? 3-ksl : 31; + SLOT->TL = (v&0x3f)*(0.75/EG_STEP); /* 0.75db step */ + + if( !(OPL->mode&0x80) ) + { /* not CSM latch total level */ + SLOT->TLL = SLOT->TL + (CH->ksl_base>>SLOT->ksl); + } +} + +/* set attack rate & decay rate */ +INLINE void set_ar_dr(FM_OPL *OPL,int slot,int v) +{ + OPL_CH *CH = &OPL->P_CH[slot/2]; + OPL_SLOT *SLOT = &CH->SLOT[slot&1]; + int ar = v>>4; + int dr = v&0x0f; + + SLOT->AR = ar ? &OPL->AR_TABLE[ar<<2] : RATE_0; + SLOT->evsa = SLOT->AR[SLOT->ksr]; + if( SLOT->evm == ENV_MOD_AR ) SLOT->evs = SLOT->evsa; + + SLOT->DR = dr ? &OPL->DR_TABLE[dr<<2] : RATE_0; + SLOT->evsd = SLOT->DR[SLOT->ksr]; + if( SLOT->evm == ENV_MOD_DR ) SLOT->evs = SLOT->evsd; +} + +/* set sustain level & release rate */ +INLINE void set_sl_rr(FM_OPL *OPL,int slot,int v) +{ + OPL_CH *CH = &OPL->P_CH[slot/2]; + OPL_SLOT *SLOT = &CH->SLOT[slot&1]; + int sl = v>>4; + int rr = v & 0x0f; + + SLOT->SL = SL_TABLE[sl]; + if( SLOT->evm == ENV_MOD_DR ) SLOT->eve = SLOT->SL; + SLOT->RR = &OPL->DR_TABLE[rr<<2]; + SLOT->evsr = SLOT->RR[SLOT->ksr]; + if( SLOT->evm == ENV_MOD_RR ) SLOT->evs = SLOT->evsr; +} + +/* operator output calcrator */ +#define OP_OUT(slot,env,con) slot->wavetable[((slot->Cnt+con)/(0x1000000/SIN_ENT))&(SIN_ENT-1)][env] +/* ---------- calcrate one of channel ---------- */ +INLINE void OPL_CALC_CH( OPL_CH *CH ) +{ + UINT32 env_out; + OPL_SLOT *SLOT; + + feedback2 = 0; + /* SLOT 1 */ + SLOT = &CH->SLOT[SLOT1]; + env_out=OPL_CALC_SLOT(SLOT); + if( env_out < EG_ENT-1 ) + { + /* PG */ + if(SLOT->vib) SLOT->Cnt += (SLOT->Incr*vib/VIB_RATE); + else SLOT->Cnt += SLOT->Incr; + /* connectoion */ + if(CH->FB) + { + int feedback1 = (CH->op1_out[0]+CH->op1_out[1])>>CH->FB; + CH->op1_out[1] = CH->op1_out[0]; + *CH->connect1 += CH->op1_out[0] = OP_OUT(SLOT,env_out,feedback1); + } + else + { + *CH->connect1 += OP_OUT(SLOT,env_out,0); + } + }else + { + CH->op1_out[1] = CH->op1_out[0]; + CH->op1_out[0] = 0; + } + /* SLOT 2 */ + SLOT = &CH->SLOT[SLOT2]; + env_out=OPL_CALC_SLOT(SLOT); + if( env_out < EG_ENT-1 ) + { + /* PG */ + if(SLOT->vib) SLOT->Cnt += (SLOT->Incr*vib/VIB_RATE); + else SLOT->Cnt += SLOT->Incr; + /* connectoion */ + outd[0] += OP_OUT(SLOT,env_out, feedback2); + } +} + +/* ---------- calcrate rythm block ---------- */ +#define WHITE_NOISE_db 6.0 +INLINE void OPL_CALC_RH( OPL_CH *CH ) +{ + UINT32 env_tam,env_sd,env_top,env_hh; + int whitenoise = (rand()&1)*(WHITE_NOISE_db/EG_STEP); + INT32 tone8; + + OPL_SLOT *SLOT; + int env_out; + + /* BD : same as FM serial mode and output level is large */ + feedback2 = 0; + /* SLOT 1 */ + SLOT = &CH[6].SLOT[SLOT1]; + env_out=OPL_CALC_SLOT(SLOT); + if( env_out < EG_ENT-1 ) + { + /* PG */ + if(SLOT->vib) SLOT->Cnt += (SLOT->Incr*vib/VIB_RATE); + else SLOT->Cnt += SLOT->Incr; + /* connectoion */ + if(CH[6].FB) + { + int feedback1 = (CH[6].op1_out[0]+CH[6].op1_out[1])>>CH[6].FB; + CH[6].op1_out[1] = CH[6].op1_out[0]; + feedback2 = CH[6].op1_out[0] = OP_OUT(SLOT,env_out,feedback1); + } + else + { + feedback2 = OP_OUT(SLOT,env_out,0); + } + }else + { + feedback2 = 0; + CH[6].op1_out[1] = CH[6].op1_out[0]; + CH[6].op1_out[0] = 0; + } + /* SLOT 2 */ + SLOT = &CH[6].SLOT[SLOT2]; + env_out=OPL_CALC_SLOT(SLOT); + if( env_out < EG_ENT-1 ) + { + /* PG */ + if(SLOT->vib) SLOT->Cnt += (SLOT->Incr*vib/VIB_RATE); + else SLOT->Cnt += SLOT->Incr; + /* connectoion */ + outd[0] += OP_OUT(SLOT,env_out, feedback2)*2; + } + + // SD (17) = mul14[fnum7] + white noise + // TAM (15) = mul15[fnum8] + // TOP (18) = fnum6(mul18[fnum8]+whitenoise) + // HH (14) = fnum7(mul18[fnum8]+whitenoise) + white noise + env_sd =OPL_CALC_SLOT(SLOT7_2) + whitenoise; + env_tam=OPL_CALC_SLOT(SLOT8_1); + env_top=OPL_CALC_SLOT(SLOT8_2); + env_hh =OPL_CALC_SLOT(SLOT7_1) + whitenoise; + + /* PG */ + if(SLOT7_1->vib) SLOT7_1->Cnt += (2*SLOT7_1->Incr*vib/VIB_RATE); + else SLOT7_1->Cnt += 2*SLOT7_1->Incr; + if(SLOT7_2->vib) SLOT7_2->Cnt += ((CH[7].fc*8)*vib/VIB_RATE); + else SLOT7_2->Cnt += (CH[7].fc*8); + if(SLOT8_1->vib) SLOT8_1->Cnt += (SLOT8_1->Incr*vib/VIB_RATE); + else SLOT8_1->Cnt += SLOT8_1->Incr; + if(SLOT8_2->vib) SLOT8_2->Cnt += ((CH[8].fc*48)*vib/VIB_RATE); + else SLOT8_2->Cnt += (CH[8].fc*48); + + tone8 = OP_OUT(SLOT8_2,whitenoise,0 ); + + /* SD */ + if( env_sd < EG_ENT-1 ) + outd[0] += OP_OUT(SLOT7_1,env_sd, 0)*8; + /* TAM */ + if( env_tam < EG_ENT-1 ) + outd[0] += OP_OUT(SLOT8_1,env_tam, 0)*2; + /* TOP-CY */ + if( env_top < EG_ENT-1 ) + outd[0] += OP_OUT(SLOT7_2,env_top,tone8)*2; + /* HH */ + if( env_hh < EG_ENT-1 ) + outd[0] += OP_OUT(SLOT7_2,env_hh,tone8)*2; +} + +/* ----------- initialize time tabls ----------- */ +static void init_timetables( FM_OPL *OPL , int ARRATE , int DRRATE ) +{ + int i; + double rate; + + /* make attack rate & decay rate tables */ + for (i = 0;i < 4;i++) OPL->AR_TABLE[i] = OPL->DR_TABLE[i] = 0; + for (i = 4;i <= 60;i++){ + rate = OPL->freqbase; /* frequency rate */ + if( i < 60 ) rate *= 1.0+(i&3)*0.25; /* b0-1 : x1 , x1.25 , x1.5 , x1.75 */ + rate *= 1<<((i>>2)-1); /* b2-5 : shift bit */ + rate *= (double)(EG_ENT<<ENV_BITS); + OPL->AR_TABLE[i] = rate / ARRATE; + OPL->DR_TABLE[i] = rate / DRRATE; + } + for (i = 60;i < 76;i++) + { + OPL->AR_TABLE[i] = EG_AED-1; + OPL->DR_TABLE[i] = OPL->DR_TABLE[60]; + } +#if 0 + for (i = 0;i < 64 ;i++){ /* make for overflow area */ + LOG(LOG_WAR,("rate %2d , ar %f ms , dr %f ms \n",i, + ((double)(EG_ENT<<ENV_BITS) / OPL->AR_TABLE[i]) * (1000.0 / OPL->rate), + ((double)(EG_ENT<<ENV_BITS) / OPL->DR_TABLE[i]) * (1000.0 / OPL->rate) )); + } +#endif +} + +/* ---------- generic table initialize ---------- */ +static int OPLOpenTable( void ) +{ + int s,t; + double rate; + int i,j; + double pom; + + /* allocate dynamic tables */ + if( (TL_TABLE = malloc(TL_MAX*2*sizeof(INT32))) == NULL) + return 0; + if( (SIN_TABLE = malloc(SIN_ENT*4 *sizeof(INT32 *))) == NULL) + { + free(TL_TABLE); + return 0; + } + if( (AMS_TABLE = malloc(AMS_ENT*2 *sizeof(INT32))) == NULL) + { + free(TL_TABLE); + free(SIN_TABLE); + return 0; + } + if( (VIB_TABLE = malloc(VIB_ENT*2 *sizeof(INT32))) == NULL) + { + free(TL_TABLE); + free(SIN_TABLE); + free(AMS_TABLE); + return 0; + } + /* make total level table */ + for (t = 0;t < EG_ENT-1 ;t++){ + rate = ((1<<TL_BITS)-1)/pow(10,EG_STEP*t/20); /* dB -> voltage */ + TL_TABLE[ t] = (int)rate; + TL_TABLE[TL_MAX+t] = -TL_TABLE[t]; +/* LOG(LOG_INF,("TotalLevel(%3d) = %x\n",t,TL_TABLE[t]));*/ + } + /* fill volume off area */ + for ( t = EG_ENT-1; t < TL_MAX ;t++){ + TL_TABLE[t] = TL_TABLE[TL_MAX+t] = 0; + } + + /* make sinwave table (total level offet) */ + /* degree 0 = degree 180 = off */ + SIN_TABLE[0] = SIN_TABLE[SIN_ENT/2] = &TL_TABLE[EG_ENT-1]; + for (s = 1;s <= SIN_ENT/4;s++){ + pom = sin(2*PI*s/SIN_ENT); /* sin */ + pom = 20*log10(1/pom); /* decibel */ + j = pom / EG_STEP; /* TL_TABLE steps */ + + /* degree 0 - 90 , degree 180 - 90 : plus section */ + SIN_TABLE[ s] = SIN_TABLE[SIN_ENT/2-s] = &TL_TABLE[j]; + /* degree 180 - 270 , degree 360 - 270 : minus section */ + SIN_TABLE[SIN_ENT/2+s] = SIN_TABLE[SIN_ENT -s] = &TL_TABLE[TL_MAX+j]; +/* LOG(LOG_INF,("sin(%3d) = %f:%f db\n",s,pom,(double)j * EG_STEP));*/ + } + for (s = 0;s < SIN_ENT;s++) + { + SIN_TABLE[SIN_ENT*1+s] = s<(SIN_ENT/2) ? SIN_TABLE[s] : &TL_TABLE[EG_ENT]; + SIN_TABLE[SIN_ENT*2+s] = SIN_TABLE[s % (SIN_ENT/2)]; + SIN_TABLE[SIN_ENT*3+s] = (s/(SIN_ENT/4))&1 ? &TL_TABLE[EG_ENT] : SIN_TABLE[SIN_ENT*2+s]; + } + + /* envelope counter -> envelope output table */ + for (i=0; i<EG_ENT; i++) + { + /* ATTACK curve */ + pom = pow( ((double)(EG_ENT-1-i)/EG_ENT) , 8 ) * EG_ENT; + /* if( pom >= EG_ENT ) pom = EG_ENT-1; */ + ENV_CURVE[i] = (int)pom; + /* DECAY ,RELEASE curve */ + ENV_CURVE[(EG_DST>>ENV_BITS)+i]= i; + } + /* off */ + ENV_CURVE[EG_OFF>>ENV_BITS]= EG_ENT-1; + /* make LFO ams table */ + for (i=0; i<AMS_ENT; i++) + { + pom = (1.0+sin(2*PI*i/AMS_ENT))/2; /* sin */ + AMS_TABLE[i] = (1.0/EG_STEP)*pom; /* 1dB */ + AMS_TABLE[AMS_ENT+i] = (4.8/EG_STEP)*pom; /* 4.8dB */ + } + /* make LFO vibrate table */ + for (i=0; i<VIB_ENT; i++) + { + /* 100cent = 1seminote = 6% ?? */ + pom = (double)VIB_RATE*0.06*sin(2*PI*i/VIB_ENT); /* +-100sect step */ + VIB_TABLE[i] = VIB_RATE + (pom*0.07); /* +- 7cent */ + VIB_TABLE[VIB_ENT+i] = VIB_RATE + (pom*0.14); /* +-14cent */ + /* LOG(LOG_INF,("vib %d=%d\n",i,VIB_TABLE[VIB_ENT+i])); */ + } + return 1; +} + + +static void OPLCloseTable( void ) +{ + free(TL_TABLE); + free(SIN_TABLE); + free(AMS_TABLE); + free(VIB_TABLE); +} + +/* CSM Key Controll */ +INLINE void CSMKeyControll(OPL_CH *CH) +{ + OPL_SLOT *slot1 = &CH->SLOT[SLOT1]; + OPL_SLOT *slot2 = &CH->SLOT[SLOT2]; + /* all key off */ + OPL_KEYOFF(slot1); + OPL_KEYOFF(slot2); + /* total level latch */ + slot1->TLL = slot1->TL + (CH->ksl_base>>slot1->ksl); + slot1->TLL = slot1->TL + (CH->ksl_base>>slot1->ksl); + /* key on */ + CH->op1_out[0] = CH->op1_out[1] = 0; + OPL_KEYON(slot1); + OPL_KEYON(slot2); +} + +/* ---------- opl initialize ---------- */ +static void OPL_initalize(FM_OPL *OPL) +{ + int fn; + + /* frequency base */ + OPL->freqbase = (OPL->rate) ? ((double)OPL->clock / OPL->rate) / 72 : 0; + /* Timer base time */ + OPL->TimerBase = 1.0/((double)OPL->clock / 72.0 ); + /* make time tables */ + init_timetables( OPL , OPL_ARRATE , OPL_DRRATE ); + /* make fnumber -> increment counter table */ + for( fn=0 ; fn < 1024 ; fn++ ) + { + OPL->FN_TABLE[fn] = OPL->freqbase * fn * FREQ_RATE * (1<<7) / 2; + } + /* LFO freq.table */ + OPL->amsIncr = OPL->rate ? (double)AMS_ENT*(1<<AMS_SHIFT) / OPL->rate * 3.7 * ((double)OPL->clock/3600000) : 0; + OPL->vibIncr = OPL->rate ? (double)VIB_ENT*(1<<VIB_SHIFT) / OPL->rate * 6.4 * ((double)OPL->clock/3600000) : 0; +} + +/* ---------- write a OPL registers ---------- */ +static void OPLWriteReg(FM_OPL *OPL, int r, int v) +{ + OPL_CH *CH; + int slot; + int block_fnum; + + switch(r&0xe0) + { + case 0x00: /* 00-1f:controll */ + switch(r&0x1f) + { + case 0x01: + /* wave selector enable */ + if(OPL->type&OPL_TYPE_WAVESEL) + { + OPL->wavesel = v&0x20; + if(!OPL->wavesel) + { + /* preset compatible mode */ + int c; + for(c=0;c<OPL->max_ch;c++) + { + OPL->P_CH[c].SLOT[SLOT1].wavetable = &SIN_TABLE[0]; + OPL->P_CH[c].SLOT[SLOT2].wavetable = &SIN_TABLE[0]; + } + } + } + return; + case 0x02: /* Timer 1 */ + OPL->T[0] = (256-v)*4; + break; + case 0x03: /* Timer 2 */ + OPL->T[1] = (256-v)*16; + return; + case 0x04: /* IRQ clear / mask and Timer enable */ + if(v&0x80) + { /* IRQ flag clear */ + OPL_STATUS_RESET(OPL,0x7f); + } + else + { /* set IRQ mask ,timer enable*/ + UINT8 st1 = v&1; + UINT8 st2 = (v>>1)&1; + /* IRQRST,T1MSK,t2MSK,EOSMSK,BRMSK,x,ST2,ST1 */ + OPL_STATUS_RESET(OPL,v&0x78); + OPL_STATUSMASK_SET(OPL,((~v)&0x78)|0x01); + /* timer 2 */ + if(OPL->st[1] != st2) + { + double interval = st2 ? (double)OPL->T[1]*OPL->TimerBase : 0.0; + OPL->st[1] = st2; + if (OPL->TimerHandler) (OPL->TimerHandler)(OPL->TimerParam+1,interval); + } + /* timer 1 */ + if(OPL->st[0] != st1) + { + double interval = st1 ? (double)OPL->T[0]*OPL->TimerBase : 0.0; + OPL->st[0] = st1; + if (OPL->TimerHandler) (OPL->TimerHandler)(OPL->TimerParam+0,interval); + } + } + return; +#if BUILD_Y8950 + case 0x06: /* Key Board OUT */ + if(OPL->type&OPL_TYPE_KEYBOARD) + { + if(OPL->keyboardhandler_w) + OPL->keyboardhandler_w(OPL->keyboard_param,v); + else + LOG(LOG_WAR,("OPL:write unmapped KEYBOARD port\n")); + } + return; + case 0x07: /* DELTA-T controll : START,REC,MEMDATA,REPT,SPOFF,x,x,RST */ + if(OPL->type&OPL_TYPE_ADPCM) + YM_DELTAT_ADPCM_Write(OPL->deltat,r-0x07,v); + return; + case 0x08: /* MODE,DELTA-T : CSM,NOTESEL,x,x,smpl,da/ad,64k,rom */ + OPL->mode = v; + v&=0x1f; /* for DELTA-T unit */ + case 0x09: /* START ADD */ + case 0x0a: + case 0x0b: /* STOP ADD */ + case 0x0c: + case 0x0d: /* PRESCALE */ + case 0x0e: + case 0x0f: /* ADPCM data */ + case 0x10: /* DELTA-N */ + case 0x11: /* DELTA-N */ + case 0x12: /* EG-CTRL */ + if(OPL->type&OPL_TYPE_ADPCM) + YM_DELTAT_ADPCM_Write(OPL->deltat,r-0x07,v); + return; +#if 0 + case 0x15: /* DAC data */ + case 0x16: + case 0x17: /* SHIFT */ + return; + case 0x18: /* I/O CTRL (Direction) */ + if(OPL->type&OPL_TYPE_IO) + OPL->portDirection = v&0x0f; + return; + case 0x19: /* I/O DATA */ + if(OPL->type&OPL_TYPE_IO) + { + OPL->portLatch = v; + if(OPL->porthandler_w) + OPL->porthandler_w(OPL->port_param,v&OPL->portDirection); + } + return; + case 0x1a: /* PCM data */ + return; +#endif +#endif + } + break; + case 0x20: /* am,vib,ksr,eg type,mul */ + slot = slot_array[r&0x1f]; + if(slot == -1) return; + set_mul(OPL,slot,v); + return; + case 0x40: + slot = slot_array[r&0x1f]; + if(slot == -1) return; + set_ksl_tl(OPL,slot,v); + return; + case 0x60: + slot = slot_array[r&0x1f]; + if(slot == -1) return; + set_ar_dr(OPL,slot,v); + return; + case 0x80: + slot = slot_array[r&0x1f]; + if(slot == -1) return; + set_sl_rr(OPL,slot,v); + return; + case 0xa0: + switch(r) + { + case 0xbd: + /* amsep,vibdep,r,bd,sd,tom,tc,hh */ + { + UINT8 rkey = OPL->rythm^v; + OPL->ams_table = &AMS_TABLE[v&0x80 ? AMS_ENT : 0]; + OPL->vib_table = &VIB_TABLE[v&0x40 ? VIB_ENT : 0]; + OPL->rythm = v&0x3f; + if(OPL->rythm&0x20) + { +#if 0 + usrintf_showmessage("OPL Rythm mode select"); +#endif + /* BD key on/off */ + if(rkey&0x10) + { + if(v&0x10) + { + OPL->P_CH[6].op1_out[0] = OPL->P_CH[6].op1_out[1] = 0; + OPL_KEYON(&OPL->P_CH[6].SLOT[SLOT1]); + OPL_KEYON(&OPL->P_CH[6].SLOT[SLOT2]); + } + else + { + OPL_KEYOFF(&OPL->P_CH[6].SLOT[SLOT1]); + OPL_KEYOFF(&OPL->P_CH[6].SLOT[SLOT2]); + } + } + /* SD key on/off */ + if(rkey&0x08) + { + if(v&0x08) OPL_KEYON(&OPL->P_CH[7].SLOT[SLOT2]); + else OPL_KEYOFF(&OPL->P_CH[7].SLOT[SLOT2]); + }/* TAM key on/off */ + if(rkey&0x04) + { + if(v&0x04) OPL_KEYON(&OPL->P_CH[8].SLOT[SLOT1]); + else OPL_KEYOFF(&OPL->P_CH[8].SLOT[SLOT1]); + } + /* TOP-CY key on/off */ + if(rkey&0x02) + { + if(v&0x02) OPL_KEYON(&OPL->P_CH[8].SLOT[SLOT2]); + else OPL_KEYOFF(&OPL->P_CH[8].SLOT[SLOT2]); + } + /* HH key on/off */ + if(rkey&0x01) + { + if(v&0x01) OPL_KEYON(&OPL->P_CH[7].SLOT[SLOT1]); + else OPL_KEYOFF(&OPL->P_CH[7].SLOT[SLOT1]); + } + } + } + return; + } + /* keyon,block,fnum */ + if( (r&0x0f) > 8) return; + CH = &OPL->P_CH[r&0x0f]; + if(!(r&0x10)) + { /* a0-a8 */ + block_fnum = (CH->block_fnum&0x1f00) | v; + } + else + { /* b0-b8 */ + int keyon = (v>>5)&1; + block_fnum = ((v&0x1f)<<8) | (CH->block_fnum&0xff); + if(CH->keyon != keyon) + { + if( (CH->keyon=keyon) ) + { + CH->op1_out[0] = CH->op1_out[1] = 0; + OPL_KEYON(&CH->SLOT[SLOT1]); + OPL_KEYON(&CH->SLOT[SLOT2]); + } + else + { + OPL_KEYOFF(&CH->SLOT[SLOT1]); + OPL_KEYOFF(&CH->SLOT[SLOT2]); + } + } + } + /* update */ + if(CH->block_fnum != block_fnum) + { + int blockRv = 7-(block_fnum>>10); + int fnum = block_fnum&0x3ff; + CH->block_fnum = block_fnum; + + CH->ksl_base = KSL_TABLE[block_fnum>>6]; + CH->fc = OPL->FN_TABLE[fnum]>>blockRv; + CH->kcode = CH->block_fnum>>9; + if( (OPL->mode&0x40) && CH->block_fnum&0x100) CH->kcode |=1; + CALC_FCSLOT(CH,&CH->SLOT[SLOT1]); + CALC_FCSLOT(CH,&CH->SLOT[SLOT2]); + } + return; + case 0xc0: + /* FB,C */ + if( (r&0x0f) > 8) return; + CH = &OPL->P_CH[r&0x0f]; + { + int feedback = (v>>1)&7; + CH->FB = feedback ? (8+1) - feedback : 0; + CH->CON = v&1; + set_algorythm(CH); + } + return; + case 0xe0: /* wave type */ + slot = slot_array[r&0x1f]; + if(slot == -1) return; + CH = &OPL->P_CH[slot/2]; + if(OPL->wavesel) + { + /* LOG(LOG_INF,("OPL SLOT %d wave select %d\n",slot,v&3)); */ + CH->SLOT[slot&1].wavetable = &SIN_TABLE[(v&0x03)*SIN_ENT]; + } + return; + } +} + +/* lock/unlock for common table */ +static int OPL_LockTable(void) +{ + num_lock++; + if(num_lock>1) return 0; + /* first time */ + cur_chip = NULL; + /* allocate total level table (128kb space) */ + if( !OPLOpenTable() ) + { + num_lock--; + return -1; + } + return 0; +} + +static void OPL_UnLockTable(void) +{ + if(num_lock) num_lock--; + if(num_lock) return; + /* last time */ + cur_chip = NULL; + OPLCloseTable(); +} + +#if (BUILD_YM3812 || BUILD_YM3526) +/*******************************************************************************/ +/* YM3812 local section */ +/*******************************************************************************/ + +/* ---------- update one of chip ----------- */ +void YM3812UpdateOne(FM_OPL *OPL, INT16 *buffer, int length) +{ + int i; + int data; + OPLSAMPLE *buf = buffer; + UINT32 amsCnt = OPL->amsCnt; + UINT32 vibCnt = OPL->vibCnt; + UINT8 rythm = OPL->rythm&0x20; + OPL_CH *CH,*R_CH; + + if( (void *)OPL != cur_chip ){ + cur_chip = (void *)OPL; + /* channel pointers */ + S_CH = OPL->P_CH; + E_CH = &S_CH[9]; + /* rythm slot */ + SLOT7_1 = &S_CH[7].SLOT[SLOT1]; + SLOT7_2 = &S_CH[7].SLOT[SLOT2]; + SLOT8_1 = &S_CH[8].SLOT[SLOT1]; + SLOT8_2 = &S_CH[8].SLOT[SLOT2]; + /* LFO state */ + amsIncr = OPL->amsIncr; + vibIncr = OPL->vibIncr; + ams_table = OPL->ams_table; + vib_table = OPL->vib_table; + } + R_CH = rythm ? &S_CH[6] : E_CH; + for( i=0; i < length ; i++ ) + { + /* channel A channel B channel C */ + /* LFO */ + ams = ams_table[(amsCnt+=amsIncr)>>AMS_SHIFT]; + vib = vib_table[(vibCnt+=vibIncr)>>VIB_SHIFT]; + outd[0] = 0; + /* FM part */ + for(CH=S_CH ; CH < R_CH ; CH++) + OPL_CALC_CH(CH); + /* Rythn part */ + if(rythm) + OPL_CALC_RH(S_CH); + /* limit check */ + data = Limit( outd[0] , OPL_MAXOUT, OPL_MINOUT ); + /* store to sound buffer */ + buf[i] = data >> OPL_OUTSB; + } + + OPL->amsCnt = amsCnt; + OPL->vibCnt = vibCnt; +#ifdef OPL_OUTPUT_LOG + if(opl_dbg_fp) + { + for(opl_dbg_chip=0;opl_dbg_chip<opl_dbg_maxchip;opl_dbg_chip++) + if( opl_dbg_opl[opl_dbg_chip] == OPL) break; + fprintf(opl_dbg_fp,"%c%c%c",0x20+opl_dbg_chip,length&0xff,length/256); + } +#endif +} +#endif /* (BUILD_YM3812 || BUILD_YM3526) */ + +#if BUILD_Y8950 + +void Y8950UpdateOne(FM_OPL *OPL, INT16 *buffer, int length) +{ + int i; + int data; + OPLSAMPLE *buf = buffer; + UINT32 amsCnt = OPL->amsCnt; + UINT32 vibCnt = OPL->vibCnt; + UINT8 rythm = OPL->rythm&0x20; + OPL_CH *CH,*R_CH; + YM_DELTAT *DELTAT = OPL->deltat; + + /* setup DELTA-T unit */ + YM_DELTAT_DECODE_PRESET(DELTAT); + + if( (void *)OPL != cur_chip ){ + cur_chip = (void *)OPL; + /* channel pointers */ + S_CH = OPL->P_CH; + E_CH = &S_CH[9]; + /* rythm slot */ + SLOT7_1 = &S_CH[7].SLOT[SLOT1]; + SLOT7_2 = &S_CH[7].SLOT[SLOT2]; + SLOT8_1 = &S_CH[8].SLOT[SLOT1]; + SLOT8_2 = &S_CH[8].SLOT[SLOT2]; + /* LFO state */ + amsIncr = OPL->amsIncr; + vibIncr = OPL->vibIncr; + ams_table = OPL->ams_table; + vib_table = OPL->vib_table; + } + R_CH = rythm ? &S_CH[6] : E_CH; + for( i=0; i < length ; i++ ) + { + /* channel A channel B channel C */ + /* LFO */ + ams = ams_table[(amsCnt+=amsIncr)>>AMS_SHIFT]; + vib = vib_table[(vibCnt+=vibIncr)>>VIB_SHIFT]; + outd[0] = 0; + /* deltaT ADPCM */ + if( DELTAT->portstate ) + YM_DELTAT_ADPCM_CALC(DELTAT); + /* FM part */ + for(CH=S_CH ; CH < R_CH ; CH++) + OPL_CALC_CH(CH); + /* Rythn part */ + if(rythm) + OPL_CALC_RH(S_CH); + /* limit check */ + data = Limit( outd[0] , OPL_MAXOUT, OPL_MINOUT ); + /* store to sound buffer */ + buf[i] = data >> OPL_OUTSB; + } + OPL->amsCnt = amsCnt; + OPL->vibCnt = vibCnt; + /* deltaT START flag */ + if( !DELTAT->portstate ) + OPL->status &= 0xfe; +} +#endif + +/* ---------- reset one of chip ---------- */ +void OPLResetChip(FM_OPL *OPL) +{ + int c,s; + int i; + + /* reset chip */ + OPL->mode = 0; /* normal mode */ + OPL_STATUS_RESET(OPL,0x7f); + /* reset with register write */ + OPLWriteReg(OPL,0x01,0); /* wabesel disable */ + OPLWriteReg(OPL,0x02,0); /* Timer1 */ + OPLWriteReg(OPL,0x03,0); /* Timer2 */ + OPLWriteReg(OPL,0x04,0); /* IRQ mask clear */ + for(i = 0xff ; i >= 0x20 ; i-- ) OPLWriteReg(OPL,i,0); + /* reset OPerator paramater */ + for( c = 0 ; c < OPL->max_ch ; c++ ) + { + OPL_CH *CH = &OPL->P_CH[c]; + /* OPL->P_CH[c].PAN = OPN_CENTER; */ + for(s = 0 ; s < 2 ; s++ ) + { + /* wave table */ + CH->SLOT[s].wavetable = &SIN_TABLE[0]; + /* CH->SLOT[s].evm = ENV_MOD_RR; */ + CH->SLOT[s].evc = EG_OFF; + CH->SLOT[s].eve = EG_OFF+1; + CH->SLOT[s].evs = 0; + } + } +#if BUILD_Y8950 + if(OPL->type&OPL_TYPE_ADPCM) + { + YM_DELTAT *DELTAT = OPL->deltat; + + DELTAT->freqbase = OPL->freqbase; + DELTAT->output_pointer = outd; + DELTAT->portshift = 5; + DELTAT->output_range = DELTAT_MIXING_LEVEL<<TL_BITS; + YM_DELTAT_ADPCM_Reset(DELTAT,0); + } +#endif +} + +/* ---------- Create one of vietual YM3812 ---------- */ +/* 'rate' is sampling rate and 'bufsiz' is the size of the */ +FM_OPL *OPLCreate(int type, int clock, int rate) +{ + char *ptr; + FM_OPL *OPL; + int state_size; + int max_ch = 9; /* normaly 9 channels */ + + if( OPL_LockTable() ==-1) return NULL; + /* allocate OPL state space */ + state_size = sizeof(FM_OPL); + state_size += sizeof(OPL_CH)*max_ch; +#if BUILD_Y8950 + if(type&OPL_TYPE_ADPCM) state_size+= sizeof(YM_DELTAT); +#endif + /* allocate memory block */ + ptr = malloc(state_size); + if(ptr==NULL) return NULL; + /* clear */ + memset(ptr,0,state_size); + OPL = (FM_OPL *)ptr; ptr+=sizeof(FM_OPL); + OPL->P_CH = (OPL_CH *)ptr; ptr+=sizeof(OPL_CH)*max_ch; +#if BUILD_Y8950 + if(type&OPL_TYPE_ADPCM) OPL->deltat = (YM_DELTAT *)ptr; ptr+=sizeof(YM_DELTAT); +#endif + /* set channel state pointer */ + OPL->type = type; + OPL->clock = clock; + OPL->rate = rate; + OPL->max_ch = max_ch; + /* init grobal tables */ + OPL_initalize(OPL); + /* reset chip */ + OPLResetChip(OPL); +#ifdef OPL_OUTPUT_LOG + if(!opl_dbg_fp) + { + opl_dbg_fp = fopen("opllog.opl","wb"); + opl_dbg_maxchip = 0; + } + if(opl_dbg_fp) + { + opl_dbg_opl[opl_dbg_maxchip] = OPL; + fprintf(opl_dbg_fp,"%c%c%c%c%c%c",0x00+opl_dbg_maxchip, + type, + clock&0xff, + (clock/0x100)&0xff, + (clock/0x10000)&0xff, + (clock/0x1000000)&0xff); + opl_dbg_maxchip++; + } +#endif + return OPL; +} + +/* ---------- Destroy one of vietual YM3812 ---------- */ +void OPLDestroy(FM_OPL *OPL) +{ +#ifdef OPL_OUTPUT_LOG + if(opl_dbg_fp) + { + fclose(opl_dbg_fp); + opl_dbg_fp = NULL; + } +#endif + OPL_UnLockTable(); + free(OPL); +} + +/* ---------- Option handlers ---------- */ + +void OPLSetTimerHandler(FM_OPL *OPL,OPL_TIMERHANDLER TimerHandler,int channelOffset) +{ + OPL->TimerHandler = TimerHandler; + OPL->TimerParam = channelOffset; +} +void OPLSetIRQHandler(FM_OPL *OPL,OPL_IRQHANDLER IRQHandler,int param) +{ + OPL->IRQHandler = IRQHandler; + OPL->IRQParam = param; +} +void OPLSetUpdateHandler(FM_OPL *OPL,OPL_UPDATEHANDLER UpdateHandler,int param) +{ + OPL->UpdateHandler = UpdateHandler; + OPL->UpdateParam = param; +} +#if BUILD_Y8950 +void OPLSetPortHandler(FM_OPL *OPL,OPL_PORTHANDLER_W PortHandler_w,OPL_PORTHANDLER_R PortHandler_r,int param) +{ + OPL->porthandler_w = PortHandler_w; + OPL->porthandler_r = PortHandler_r; + OPL->port_param = param; +} + +void OPLSetKeyboardHandler(FM_OPL *OPL,OPL_PORTHANDLER_W KeyboardHandler_w,OPL_PORTHANDLER_R KeyboardHandler_r,int param) +{ + OPL->keyboardhandler_w = KeyboardHandler_w; + OPL->keyboardhandler_r = KeyboardHandler_r; + OPL->keyboard_param = param; +} +#endif +/* ---------- YM3812 I/O interface ---------- */ +int OPLWrite(FM_OPL *OPL,int a,int v) +{ + if( !(a&1) ) + { /* address port */ + OPL->address = v & 0xff; + } + else + { /* data port */ + if(OPL->UpdateHandler) OPL->UpdateHandler(OPL->UpdateParam,0); +#ifdef OPL_OUTPUT_LOG + if(opl_dbg_fp) + { + for(opl_dbg_chip=0;opl_dbg_chip<opl_dbg_maxchip;opl_dbg_chip++) + if( opl_dbg_opl[opl_dbg_chip] == OPL) break; + fprintf(opl_dbg_fp,"%c%c%c",0x10+opl_dbg_chip,OPL->address,v); + } +#endif + OPLWriteReg(OPL,OPL->address,v); + } + return OPL->status>>7; +} + +unsigned char OPLRead(FM_OPL *OPL,int a) +{ + if( !(a&1) ) + { /* status port */ + return OPL->status & (OPL->statusmask|0x80); + } + /* data port */ + switch(OPL->address) + { + case 0x05: /* KeyBoard IN */ + if(OPL->type&OPL_TYPE_KEYBOARD) + { + if(OPL->keyboardhandler_r) + return OPL->keyboardhandler_r(OPL->keyboard_param); + else + LOG(LOG_WAR,("OPL:read unmapped KEYBOARD port\n")); + } + return 0; +#if 0 + case 0x0f: /* ADPCM-DATA */ + return 0; +#endif + case 0x19: /* I/O DATA */ + if(OPL->type&OPL_TYPE_IO) + { + if(OPL->porthandler_r) + return OPL->porthandler_r(OPL->port_param); + else + LOG(LOG_WAR,("OPL:read unmapped I/O port\n")); + } + return 0; + case 0x1a: /* PCM-DATA */ + return 0; + } + return 0; +} + +int OPLTimerOver(FM_OPL *OPL,int c) +{ + if( c ) + { /* Timer B */ + OPL_STATUS_SET(OPL,0x20); + } + else + { /* Timer A */ + OPL_STATUS_SET(OPL,0x40); + /* CSM mode key,TL controll */ + if( OPL->mode & 0x80 ) + { /* CSM mode total level latch and auto key on */ + int ch; + if(OPL->UpdateHandler) OPL->UpdateHandler(OPL->UpdateParam,0); + for(ch=0;ch<9;ch++) + CSMKeyControll( &OPL->P_CH[ch] ); + } + } + /* reload timer */ + if (OPL->TimerHandler) (OPL->TimerHandler)(OPL->TimerParam+c,(double)OPL->T[c]*OPL->TimerBase); + return OPL->status>>7; +} diff --git a/hw/fmopl.h b/hw/fmopl.h new file mode 100644 index 0000000000..a01ff902c7 --- /dev/null +++ b/hw/fmopl.h @@ -0,0 +1,174 @@ +#ifndef __FMOPL_H_ +#define __FMOPL_H_ + +/* --- select emulation chips --- */ +#define BUILD_YM3812 (HAS_YM3812) +//#define BUILD_YM3526 (HAS_YM3526) +//#define BUILD_Y8950 (HAS_Y8950) + +/* --- system optimize --- */ +/* select bit size of output : 8 or 16 */ +#define OPL_OUTPUT_BIT 16 + +/* compiler dependence */ +#ifndef OSD_CPU_H +#define OSD_CPU_H +typedef unsigned char UINT8; /* unsigned 8bit */ +typedef unsigned short UINT16; /* unsigned 16bit */ +typedef unsigned int UINT32; /* unsigned 32bit */ +typedef signed char INT8; /* signed 8bit */ +typedef signed short INT16; /* signed 16bit */ +typedef signed int INT32; /* signed 32bit */ +#endif + +#if (OPL_OUTPUT_BIT==16) +typedef INT16 OPLSAMPLE; +#endif +#if (OPL_OUTPUT_BIT==8) +typedef unsigned char OPLSAMPLE; +#endif + + +#if BUILD_Y8950 +#include "ymdeltat.h" +#endif + +typedef void (*OPL_TIMERHANDLER)(int channel,double interval_Sec); +typedef void (*OPL_IRQHANDLER)(int param,int irq); +typedef void (*OPL_UPDATEHANDLER)(int param,int min_interval_us); +typedef void (*OPL_PORTHANDLER_W)(int param,unsigned char data); +typedef unsigned char (*OPL_PORTHANDLER_R)(int param); + +/* !!!!! here is private section , do not access there member direct !!!!! */ + +#define OPL_TYPE_WAVESEL 0x01 /* waveform select */ +#define OPL_TYPE_ADPCM 0x02 /* DELTA-T ADPCM unit */ +#define OPL_TYPE_KEYBOARD 0x04 /* keyboard interface */ +#define OPL_TYPE_IO 0x08 /* I/O port */ + +/* Saving is necessary for member of the 'R' mark for suspend/resume */ +/* ---------- OPL one of slot ---------- */ +typedef struct fm_opl_slot { + INT32 TL; /* total level :TL << 8 */ + INT32 TLL; /* adjusted now TL */ + UINT8 KSR; /* key scale rate :(shift down bit) */ + INT32 *AR; /* attack rate :&AR_TABLE[AR<<2] */ + INT32 *DR; /* decay rate :&DR_TALBE[DR<<2] */ + INT32 SL; /* sustin level :SL_TALBE[SL] */ + INT32 *RR; /* release rate :&DR_TABLE[RR<<2] */ + UINT8 ksl; /* keyscale level :(shift down bits) */ + UINT8 ksr; /* key scale rate :kcode>>KSR */ + UINT32 mul; /* multiple :ML_TABLE[ML] */ + UINT32 Cnt; /* frequency count : */ + UINT32 Incr; /* frequency step : */ + /* envelope generator state */ + UINT8 eg_typ; /* envelope type flag */ + UINT8 evm; /* envelope phase */ + INT32 evc; /* envelope counter */ + INT32 eve; /* envelope counter end point */ + INT32 evs; /* envelope counter step */ + INT32 evsa; /* envelope step for AR :AR[ksr] */ + INT32 evsd; /* envelope step for DR :DR[ksr] */ + INT32 evsr; /* envelope step for RR :RR[ksr] */ + /* LFO */ + UINT8 ams; /* ams flag */ + UINT8 vib; /* vibrate flag */ + /* wave selector */ + INT32 **wavetable; +}OPL_SLOT; + +/* ---------- OPL one of channel ---------- */ +typedef struct fm_opl_channel { + OPL_SLOT SLOT[2]; + UINT8 CON; /* connection type */ + UINT8 FB; /* feed back :(shift down bit) */ + INT32 *connect1; /* slot1 output pointer */ + INT32 *connect2; /* slot2 output pointer */ + INT32 op1_out[2]; /* slot1 output for selfeedback */ + /* phase generator state */ + UINT32 block_fnum; /* block+fnum : */ + UINT8 kcode; /* key code : KeyScaleCode */ + UINT32 fc; /* Freq. Increment base */ + UINT32 ksl_base; /* KeyScaleLevel Base step */ + UINT8 keyon; /* key on/off flag */ +} OPL_CH; + +/* OPL state */ +typedef struct fm_opl_f { + UINT8 type; /* chip type */ + int clock; /* master clock (Hz) */ + int rate; /* sampling rate (Hz) */ + double freqbase; /* frequency base */ + double TimerBase; /* Timer base time (==sampling time) */ + UINT8 address; /* address register */ + UINT8 status; /* status flag */ + UINT8 statusmask; /* status mask */ + UINT32 mode; /* Reg.08 : CSM , notesel,etc. */ + /* Timer */ + int T[2]; /* timer counter */ + UINT8 st[2]; /* timer enable */ + /* FM channel slots */ + OPL_CH *P_CH; /* pointer of CH */ + int max_ch; /* maximum channel */ + /* Rythm sention */ + UINT8 rythm; /* Rythm mode , key flag */ +#if BUILD_Y8950 + /* Delta-T ADPCM unit (Y8950) */ + YM_DELTAT *deltat; /* DELTA-T ADPCM */ +#endif + /* Keyboard / I/O interface unit (Y8950) */ + UINT8 portDirection; + UINT8 portLatch; + OPL_PORTHANDLER_R porthandler_r; + OPL_PORTHANDLER_W porthandler_w; + int port_param; + OPL_PORTHANDLER_R keyboardhandler_r; + OPL_PORTHANDLER_W keyboardhandler_w; + int keyboard_param; + /* time tables */ + INT32 AR_TABLE[75]; /* atttack rate tables */ + INT32 DR_TABLE[75]; /* decay rate tables */ + UINT32 FN_TABLE[1024]; /* fnumber -> increment counter */ + /* LFO */ + INT32 *ams_table; + INT32 *vib_table; + INT32 amsCnt; + INT32 amsIncr; + INT32 vibCnt; + INT32 vibIncr; + /* wave selector enable flag */ + UINT8 wavesel; + /* external event callback handler */ + OPL_TIMERHANDLER TimerHandler; /* TIMER handler */ + int TimerParam; /* TIMER parameter */ + OPL_IRQHANDLER IRQHandler; /* IRQ handler */ + int IRQParam; /* IRQ parameter */ + OPL_UPDATEHANDLER UpdateHandler; /* stream update handler */ + int UpdateParam; /* stream update parameter */ +} FM_OPL; + +/* ---------- Generic interface section ---------- */ +#define OPL_TYPE_YM3526 (0) +#define OPL_TYPE_YM3812 (OPL_TYPE_WAVESEL) +#define OPL_TYPE_Y8950 (OPL_TYPE_ADPCM|OPL_TYPE_KEYBOARD|OPL_TYPE_IO) + +FM_OPL *OPLCreate(int type, int clock, int rate); +void OPLDestroy(FM_OPL *OPL); +void OPLSetTimerHandler(FM_OPL *OPL,OPL_TIMERHANDLER TimerHandler,int channelOffset); +void OPLSetIRQHandler(FM_OPL *OPL,OPL_IRQHANDLER IRQHandler,int param); +void OPLSetUpdateHandler(FM_OPL *OPL,OPL_UPDATEHANDLER UpdateHandler,int param); +/* Y8950 port handlers */ +void OPLSetPortHandler(FM_OPL *OPL,OPL_PORTHANDLER_W PortHandler_w,OPL_PORTHANDLER_R PortHandler_r,int param); +void OPLSetKeyboardHandler(FM_OPL *OPL,OPL_PORTHANDLER_W KeyboardHandler_w,OPL_PORTHANDLER_R KeyboardHandler_r,int param); + +void OPLResetChip(FM_OPL *OPL); +int OPLWrite(FM_OPL *OPL,int a,int v); +unsigned char OPLRead(FM_OPL *OPL,int a); +int OPLTimerOver(FM_OPL *OPL,int c); + +/* YM3626/YM3812 local section */ +void YM3812UpdateOne(FM_OPL *OPL, INT16 *buffer, int length); + +void Y8950UpdateOne(FM_OPL *OPL, INT16 *buffer, int length); + +#endif @@ -554,13 +554,10 @@ void pc_init(int ram_size, int vga_ram_size, int boot_device, kbd_init(); DMA_init(0); -#ifndef _WIN32 if (audio_enabled) { /* no audio supported yet for win32 */ AUD_init(); - SB16_init(); } -#endif floppy_controller = fdctrl_init(6, 2, 0, 0x3f0, fd_table); @@ -1,7 +1,7 @@ /* * QEMU Soundblaster 16 emulation * - * Copyright (c) 2003 Vassili Karpov (malc) + * Copyright (c) 2003-2004 Vassili Karpov (malc) * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal @@ -23,32 +23,20 @@ */ #include "vl.h" -#define MIN(a, b) ((a)>(b)?(b):(a)) -#define LENOFA(a) ((int) (sizeof(a)/sizeof(a[0]))) - -#define dolog(...) fprintf (stderr, "sb16: " __VA_ARGS__); +/* #define DEBUG */ +#define AUDIO_CAP "sb16" +#include "audio/audio.h" -/* #define DEBUG_SB16 */ +#define LENOFA(a) ((int) (sizeof(a)/sizeof(a[0]))) -#ifdef DEBUG_SB16 -#define lwarn(...) fprintf (stderr, "sb16: " __VA_ARGS__) -#define linfo(...) fprintf (stderr, "sb16: " __VA_ARGS__) -#define ldebug(...) fprintf (stderr, "sb16: " __VA_ARGS__) -#else -#define lwarn(...) -#define linfo(...) -#define ldebug(...) -#endif +/* #define DEBUG_SB16_MOST */ -#define IO_READ_PROTO(name) \ +#define IO_READ_PROTO(name) \ uint32_t name (void *opaque, uint32_t nport) -#define IO_WRITE_PROTO(name) \ +#define IO_WRITE_PROTO(name) \ void name (void *opaque, uint32_t nport, uint32_t val) -static const char e3[] = - "COPYRIGHT (C) CREATIVE TECHNOLOGY LTD, 1992\0" - "COPYRIGHT (C) CREATIVE TECHNOLOGY LTD, 1994-1997"; - /* "COPYRIGHT (C) CREATIVE TECHNOLOGY LTD, 1994."; */ +static const char e3[] = "COPYRIGHT (C) CREATIVE TECHNOLOGY LTD, 1992."; static struct { int ver_lo; @@ -57,38 +45,58 @@ static struct { int dma; int hdma; int port; - int mix_block; -} sb = {5, 4, 5, 1, 5, 0x220, -1}; - -static int mix_block, noirq; +} conf = {5, 4, 5, 1, 5, 0x220}; typedef struct SB16State { + int irq; + int dma; + int hdma; + int port; + int ver; + int in_index; int out_data_len; int fmt_stereo; int fmt_signed; int fmt_bits; + audfmt_e fmt; int dma_auto; - int dma_buffer_size; + int block_size; int fifo; int freq; int time_const; int speaker; int needed_bytes; int cmd; - int dma_pos; int use_hdma; + int highspeed; + int can_write; int v2x6; + uint8_t csp_param; + uint8_t csp_value; + uint8_t csp_mode; + uint8_t csp_regs[256]; + uint8_t csp_index; + uint8_t csp_reg83[4]; + int csp_reg83r; + int csp_reg83w; + uint8_t in2_data[10]; - uint8_t out_data[1024]; + uint8_t out_data[50]; + uint8_t test_reg; + uint8_t last_read_byte; + int nzero; int left_till_irq; - uint64_t nzero; - uint8_t last_read_byte; - uint8_t test_reg; + int dma_running; + int bytes_per_second; + int align; + SWVoice *voice; + + QEMUTimer *ts, *aux_ts; /* mixer state */ int mixer_nreg; uint8_t mixer_regs[256]; @@ -97,664 +105,853 @@ typedef struct SB16State { /* XXX: suppress that and use a context */ static struct SB16State dsp; +static int magic_of_irq (int irq) +{ + switch (irq) { + case 5: + return 2; + case 7: + return 4; + case 9: + return 1; + case 10: + return 8; + default: + dolog ("bad irq %d\n", irq); + return 2; + } +} + +static int irq_of_magic (int magic) +{ + switch (magic) { + case 1: + return 9; + case 2: + return 5; + case 4: + return 7; + case 8: + return 10; + default: + dolog ("bad irq magic %d\n", magic); + return -1; + } +} + +#if 0 static void log_dsp (SB16State *dsp) { - ldebug ("%c:%c:%d:%c:dmabuf=%d:pos=%d:freq=%d:timeconst=%d:speaker=%d\n", - dsp->fmt_stereo ? 'S' : 'M', - dsp->fmt_signed ? 'S' : 'U', - dsp->fmt_bits, - dsp->dma_auto ? 'a' : 's', - dsp->dma_buffer_size, - dsp->dma_pos, - dsp->freq, - dsp->time_const, - dsp->speaker); + ldebug ("%s:%s:%d:%s:dmasize=%d:freq=%d:const=%d:speaker=%d\n", + dsp->fmt_stereo ? "Stereo" : "Mono", + dsp->fmt_signed ? "Signed" : "Unsigned", + dsp->fmt_bits, + dsp->dma_auto ? "Auto" : "Single", + dsp->block_size, + dsp->freq, + dsp->time_const, + dsp->speaker); +} +#endif + +static void speaker (SB16State *s, int on) +{ + s->speaker = on; + /* AUD_enable (s->voice, on); */ } -static void control (int hold) +static void control (SB16State *s, int hold) { - linfo ("%d high %d\n", hold, dsp.use_hdma); + int dma = s->use_hdma ? s->hdma : s->dma; + s->dma_running = hold; + + ldebug ("hold %d high %d dma %d\n", hold, s->use_hdma, dma); + if (hold) { - if (dsp.use_hdma) - DMA_hold_DREQ (sb.hdma); - else - DMA_hold_DREQ (sb.dma); + DMA_hold_DREQ (dma); + AUD_enable (s->voice, 1); } else { - if (dsp.use_hdma) - DMA_release_DREQ (sb.hdma); - else - DMA_release_DREQ (sb.dma); + DMA_release_DREQ (dma); + AUD_enable (s->voice, 0); } } -static void dma_cmd (uint8_t cmd, uint8_t d0, int dma_len) +static void aux_timer (void *opaque) { - int bps; - audfmt_e fmt; + SB16State *s = opaque; + s->can_write = 1; + pic_set_irq (s->irq, 1); +} + +#define DMA8_AUTO 1 +#define DMA8_HIGH 2 + +static void dma_cmd8 (SB16State *s, int mask, int dma_len) +{ + s->fmt = AUD_FMT_U8; + s->use_hdma = 0; + s->fmt_bits = 8; + s->fmt_signed = 0; + s->fmt_stereo = (s->mixer_regs[0x0e] & 2) != 0; + if (-1 == s->time_const) { + s->freq = 11025; + } + else { + int tmp = (256 - s->time_const); + s->freq = (1000000 + (tmp / 2)) / tmp; + } + + if (-1 != dma_len) + s->block_size = dma_len + 1; + + s->freq >>= s->fmt_stereo; + s->left_till_irq = s->block_size; + s->bytes_per_second = (s->freq << s->fmt_stereo); + /* s->highspeed = (mask & DMA8_HIGH) != 0; */ + s->dma_auto = (mask & DMA8_AUTO) != 0; + s->align = (1 << s->fmt_stereo) - 1; + + ldebug ("freq %d, stereo %d, sign %d, bits %d, " + "dma %d, auto %d, fifo %d, high %d\n", + s->freq, s->fmt_stereo, s->fmt_signed, s->fmt_bits, + s->block_size, s->dma_auto, s->fifo, s->highspeed); + + if (s->freq) + s->voice = AUD_open (s->voice, "sb16", s->freq, + 1 << s->fmt_stereo, s->fmt); + + control (s, 1); + speaker (s, 1); +} - dsp.use_hdma = cmd < 0xc0; - dsp.fifo = (cmd >> 1) & 1; - dsp.dma_auto = (cmd >> 2) & 1; +static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len) +{ + s->use_hdma = cmd < 0xc0; + s->fifo = (cmd >> 1) & 1; + s->dma_auto = (cmd >> 2) & 1; + s->fmt_signed = (d0 >> 4) & 1; + s->fmt_stereo = (d0 >> 5) & 1; switch (cmd >> 4) { case 11: - dsp.fmt_bits = 16; + s->fmt_bits = 16; break; case 12: - dsp.fmt_bits = 8; + s->fmt_bits = 8; break; } - dsp.fmt_signed = (d0 >> 4) & 1; - dsp.fmt_stereo = (d0 >> 5) & 1; - - if (-1 != dsp.time_const) { - int tmp; - - tmp = 256 - dsp.time_const; - dsp.freq = (1000000 + (tmp / 2)) / tmp; + if (-1 != s->time_const) { +#if 1 + int tmp = 256 - s->time_const; + s->freq = (1000000 + (tmp / 2)) / tmp; +#else + /* s->freq = 1000000 / ((255 - s->time_const) << s->fmt_stereo); */ + s->freq = 1000000 / ((255 - s->time_const)); +#endif + s->time_const = -1; } - bps = 1 << (16 == dsp.fmt_bits); - if (-1 != dma_len) - dsp.dma_buffer_size = (dma_len + 1) * bps; + s->block_size = dma_len + 1; + s->block_size <<= (s->fmt_bits == 16); + if (!s->dma_auto) /* Miles Sound System ? */ + s->block_size <<= s->fmt_stereo; - linfo ("frequency %d, stereo %d, signed %d, bits %d, size %d, auto %d\n", - dsp.freq, dsp.fmt_stereo, dsp.fmt_signed, dsp.fmt_bits, - dsp.dma_buffer_size, dsp.dma_auto); + ldebug ("freq %d, stereo %d, sign %d, bits %d, " + "dma %d, auto %d, fifo %d, high %d\n", + s->freq, s->fmt_stereo, s->fmt_signed, s->fmt_bits, + s->block_size, s->dma_auto, s->fifo, s->highspeed); - if (16 == dsp.fmt_bits) { - if (dsp.fmt_signed) { - fmt = AUD_FMT_S16; + if (16 == s->fmt_bits) { + if (s->fmt_signed) { + s->fmt = AUD_FMT_S16; } else { - fmt = AUD_FMT_U16; + s->fmt = AUD_FMT_U16; } } else { - if (dsp.fmt_signed) { - fmt = AUD_FMT_S8; + if (s->fmt_signed) { + s->fmt = AUD_FMT_S8; } else { - fmt = AUD_FMT_U8; + s->fmt = AUD_FMT_U8; } } - dsp.dma_pos = 0; - dsp.left_till_irq = dsp.dma_buffer_size; + s->left_till_irq = s->block_size; - if (sb.mix_block) { - mix_block = sb.mix_block; - } - else { - int align; + s->bytes_per_second = (s->freq << s->fmt_stereo) << (s->fmt_bits == 16); + s->highspeed = 0; + s->align = (1 << (s->fmt_stereo + (s->fmt_bits == 16))) - 1; - align = bps << dsp.fmt_stereo; - mix_block = ((dsp.freq * align) / 100) & ~(align - 1); - } + if (s->freq) + s->voice = AUD_open (s->voice, "sb16", s->freq, + 1 << s->fmt_stereo, s->fmt); - if (dsp.freq) - AUD_reset (dsp.freq, 1 << dsp.fmt_stereo, fmt); - control (1); - dsp.speaker = 1; + control (s, 1); + speaker (s, 1); } -static inline void dsp_out_data(SB16State *dsp, int val) +static inline void dsp_out_data (SB16State *s, uint8_t val) { - if (dsp->out_data_len < sizeof(dsp->out_data)) - dsp->out_data[dsp->out_data_len++] = val; + ldebug ("outdata %#x\n", val); + if (s->out_data_len < sizeof (s->out_data)) + s->out_data[s->out_data_len++] = val; } -static inline uint8_t dsp_get_data(SB16State *dsp) +static inline uint8_t dsp_get_data (SB16State *s) { - if (dsp->in_index) - return dsp->in2_data[--dsp->in_index]; - else + if (s->in_index) + return s->in2_data[--s->in_index]; + else { + dolog ("buffer underflow\n"); return 0; + } } -static void command (SB16State *dsp, uint8_t cmd) +static void command (SB16State *s, uint8_t cmd) { - linfo ("command: %#x\n", cmd); + ldebug ("command %#x\n", cmd); if (cmd > 0xaf && cmd < 0xd0) { - if (cmd & 8) - goto error; + if (cmd & 8) { + dolog ("ADC not yet supported (command %#x)\n", cmd); + } switch (cmd >> 4) { case 11: case 12: break; default: - dolog ("command: %#x wrong bits specification\n", cmd); - goto error; + dolog ("%#x wrong bits\n", cmd); } - dsp->needed_bytes = 3; + s->needed_bytes = 3; } else { switch (cmd) { - case 0x00: - case 0xe7: - /* IMS uses those when probing for sound devices */ - return; - case 0x03: + dsp_out_data (s, 0x10); /* s->csp_param); */ + goto warn; + case 0x04: - dsp_out_data (dsp, 0); - return; + s->needed_bytes = 1; + goto warn; case 0x05: - dsp->needed_bytes = 2; - break; + s->needed_bytes = 2; + goto warn; + + case 0x08: + /* __asm__ ("int3"); */ + goto warn; case 0x0e: - dsp->needed_bytes = 2; - break; + s->needed_bytes = 2; + goto warn; + + case 0x09: + dsp_out_data (s, 0xf8); + goto warn; case 0x0f: - dsp->needed_bytes = 1; - break; + s->needed_bytes = 1; + goto warn; case 0x10: - dsp->needed_bytes = 1; - break; + s->needed_bytes = 1; + goto warn; case 0x14: - dsp->needed_bytes = 2; - dsp->dma_buffer_size = 0; + s->needed_bytes = 2; + s->block_size = 0; break; - case 0x20: - dsp_out_data(dsp, 0xff); - break; + case 0x20: /* Direct ADC, Juice/PL */ + dsp_out_data (s, 0xff); + goto warn; case 0x35: - lwarn ("MIDI commands not implemented\n"); + dolog ("MIDI command(0x35) not implemented\n"); break; case 0x40: - dsp->freq = -1; - dsp->time_const = -1; - dsp->needed_bytes = 1; + s->freq = -1; + s->time_const = -1; + s->needed_bytes = 1; break; case 0x41: - case 0x42: - dsp->freq = -1; - dsp->time_const = -1; - dsp->needed_bytes = 2; + s->freq = -1; + s->time_const = -1; + s->needed_bytes = 2; break; + case 0x42: + s->freq = -1; + s->time_const = -1; + s->needed_bytes = 2; + goto warn; + case 0x45: - dsp_out_data (dsp, 0xaa); + dsp_out_data (s, 0xaa); + goto warn; + case 0x47: /* Continue Auto-Initialize DMA 16bit */ break; case 0x48: - dsp->needed_bytes = 2; + s->needed_bytes = 2; break; - case 0x27: /* ????????? */ - case 0x4e: - return; - case 0x80: - cmd = -1; + s->needed_bytes = 2; break; case 0x90: case 0x91: - { - uint8_t d0; - - d0 = 4; - /* if (dsp->fmt_signed) d0 |= 16; */ - /* if (dsp->fmt_stereo) d0 |= 32; */ - dma_cmd (cmd == 0x90 ? 0xc4 : 0xc0, d0, -1); - cmd = -1; - break; - } + dma_cmd8 (s, ((cmd & 1) == 0) | DMA8_HIGH, -1); + break; - case 0xd0: /* XXX */ - control (0); - return; + case 0xd0: /* halt DMA operation. 8bit */ + control (s, 0); + break; - case 0xd1: - dsp->speaker = 1; + case 0xd1: /* speaker on */ + speaker (s, 1); break; - case 0xd3: - dsp->speaker = 0; - return; + case 0xd3: /* speaker off */ + speaker (s, 0); + break; - case 0xd4: - control (1); + case 0xd4: /* continue DMA operation. 8bit */ + control (s, 1); break; - case 0xd5: - control (0); + case 0xd5: /* halt DMA operation. 16bit */ + control (s, 0); break; - case 0xd6: - control (1); + case 0xd6: /* continue DMA operation. 16bit */ + control (s, 1); break; - case 0xd9: - control (0); - dsp->dma_auto = 0; - return; + case 0xd9: /* exit auto-init DMA after this block. 16bit */ + s->dma_auto = 0; + break; - case 0xda: - control (0); - dsp->dma_auto = 0; + case 0xda: /* exit auto-init DMA after this block. 8bit */ + s->dma_auto = 0; break; case 0xe0: - dsp->needed_bytes = 1; - break; + s->needed_bytes = 1; + goto warn; case 0xe1: - dsp_out_data(dsp, sb.ver_lo); - dsp_out_data(dsp, sb.ver_hi); - return; + dsp_out_data (s, s->ver & 0xff); + dsp_out_data (s, s->ver >> 8); + break; + + case 0xe2: + s->needed_bytes = 1; + goto warn; case 0xe3: { int i; - for (i = sizeof (e3) - 1; i >= 0; i--) - dsp_out_data (dsp, e3[i]); - return; + for (i = sizeof (e3) - 1; i >= 0; --i) + dsp_out_data (s, e3[i]); } + break; case 0xe4: /* write test reg */ - dsp->needed_bytes = 1; + s->needed_bytes = 1; break; + case 0xe7: + dolog ("Attempt to probe for ESS (0xe7)?\n"); + return; + case 0xe8: /* read test reg */ - dsp_out_data (dsp, dsp->test_reg); + dsp_out_data (s, s->test_reg); break; case 0xf2: - dsp_out_data (dsp, 0xaa); - dsp->mixer_regs[0x82] |= dsp->mixer_regs[0x80]; - pic_set_irq (sb.irq, 1); - return; + case 0xf3: + dsp_out_data (s, 0xaa); + s->mixer_regs[0x82] |= (cmd == 0xf2) ? 1 : 2; + pic_set_irq (s->irq, 1); + break; case 0xf9: - dsp->needed_bytes = 1; - break; + s->needed_bytes = 1; + goto warn; case 0xfa: - dsp_out_data (dsp, 0); - break; + dsp_out_data (s, 0); + goto warn; case 0xfc: /* FIXME */ - dsp_out_data (dsp, 0); - break; + dsp_out_data (s, 0); + goto warn; default: dolog ("unrecognized command %#x\n", cmd); - goto error; + return; } } - dsp->cmd = cmd; + + s->cmd = cmd; + if (!s->needed_bytes) + ldebug ("\n"); return; - error: + warn: + dolog ("warning command %#x,%d is not trully understood yet\n", + cmd, s->needed_bytes); + s->cmd = cmd; return; } -static void complete (SB16State *dsp) +static uint16_t dsp_get_lohi (SB16State *s) +{ + uint8_t hi = dsp_get_data (s); + uint8_t lo = dsp_get_data (s); + return (hi << 8) | lo; +} + +static uint16_t dsp_get_hilo (SB16State *s) +{ + uint8_t lo = dsp_get_data (s); + uint8_t hi = dsp_get_data (s); + return (hi << 8) | lo; +} + +static void complete (SB16State *s) { int d0, d1, d2; - linfo ("complete command %#x, in_index %d, needed_bytes %d\n", - dsp->cmd, dsp->in_index, dsp->needed_bytes); + ldebug ("complete command %#x, in_index %d, needed_bytes %d\n", + s->cmd, s->in_index, s->needed_bytes); - if (dsp->cmd > 0xaf && dsp->cmd < 0xd0) { - d2 = dsp_get_data (dsp); - d1 = dsp_get_data (dsp); - d0 = dsp_get_data (dsp); + if (s->cmd > 0xaf && s->cmd < 0xd0) { + d2 = dsp_get_data (s); + d1 = dsp_get_data (s); + d0 = dsp_get_data (s); - ldebug ("d0 = %d, d1 = %d, d2 = %d\n", - d0, d1, d2); - dma_cmd (dsp->cmd, d0, d1 + (d2 << 8)); + if (s->cmd & 8) { + dolog ("ADC params cmd = %#x d0 = %d, d1 = %d, d2 = %d\n", + s->cmd, d0, d1, d2); + } + else { + ldebug ("cmd = %#x d0 = %d, d1 = %d, d2 = %d\n", + s->cmd, d0, d1, d2); + dma_cmd (s, s->cmd, d0, d1 + (d2 << 8)); + } } else { - switch (dsp->cmd) { + switch (s->cmd) { case 0x04: - case 0x10: - dsp_get_data (dsp); + s->csp_mode = dsp_get_data (s); + s->csp_reg83r = 0; + s->csp_reg83w = 0; + ldebug ("CSP command 0x04: mode=%#x\n", s->csp_mode); break; - case 0x0f: - d0 = dsp_get_data (dsp); - dsp_out_data (dsp, 0xf8); + case 0x05: + s->csp_param = dsp_get_data (s); + s->csp_value = dsp_get_data (s); + ldebug ("CSP command 0x05: param=%#x value=%#x\n", + s->csp_param, + s->csp_value); break; - case 0x05: case 0x0e: - dsp_get_data (dsp); - dsp_get_data (dsp); + d0 = dsp_get_data (s); + d1 = dsp_get_data (s); + ldebug ("write CSP register %d <- %#x\n", d1, d0); + if (d1 == 0x83) { + ldebug ("0x83[%d] <- %#x\n", s->csp_reg83r, d0); + s->csp_reg83[s->csp_reg83r % 4] = d0; + s->csp_reg83r += 1; + } + else + s->csp_regs[d1] = d0; break; - case 0x14: - { - int save_left; - int save_pos; - - d1 = dsp_get_data (dsp); - d0 = dsp_get_data (dsp); + case 0x0f: + d0 = dsp_get_data (s); + ldebug ("read CSP register %#x -> %#x, mode=%#x\n", + d0, s->csp_regs[d0], s->csp_mode); + if (d0 == 0x83) { + ldebug ("0x83[%d] -> %#x\n", + s->csp_reg83w, + s->csp_reg83[s->csp_reg83w % 4]); + dsp_out_data (s, s->csp_reg83[s->csp_reg83w % 4]); + s->csp_reg83w += 1; + } + else + dsp_out_data (s, s->csp_regs[d0]); + break; - save_left = dsp->left_till_irq; - save_pos = dsp->dma_pos; - dma_cmd (0xc0, 0, d0 + (d1 << 8)); - dsp->left_till_irq = save_left; - dsp->dma_pos = save_pos; + case 0x10: + d0 = dsp_get_data (s); + dolog ("cmd 0x10 d0=%#x\n", d0); + break; - linfo ("set buffer size data[%d, %d] %d pos %d\n", - d0, d1, dsp->dma_buffer_size, dsp->dma_pos); - break; - } + case 0x14: + dma_cmd8 (s, 0, dsp_get_lohi (s)); + /* s->can_write = 0; */ + /* qemu_mod_timer (s->aux_ts, qemu_get_clock (vm_clock) + (ticks_per_sec * 320) / 1000000); */ + break; case 0x40: - dsp->time_const = dsp_get_data (dsp); - linfo ("set time const %d\n", dsp->time_const); + s->time_const = dsp_get_data (s); + ldebug ("set time const %d\n", s->time_const); break; - case 0x41: - case 0x42: - d1 = dsp_get_data (dsp); - d0 = dsp_get_data (dsp); + case 0x42: /* FT2 sets output freq with this, go figure */ + dolog ("cmd 0x42 might not do what it think it should\n"); - dsp->freq = d1 + (d0 << 8); - linfo ("set freq %#x, %#x = %d\n", d1, d0, dsp->freq); + case 0x41: + s->freq = dsp_get_hilo (s); + ldebug ("set freq %d\n", s->freq); break; case 0x48: - d1 = dsp_get_data (dsp); - d0 = dsp_get_data (dsp); - dsp->dma_buffer_size = d1 + (d0 << 8); - linfo ("set dma len %#x, %#x = %d\n", - d1, d0, dsp->dma_buffer_size); + s->block_size = dsp_get_lohi (s); + /* s->highspeed = 1; */ + ldebug ("set dma block len %d\n", s->block_size); + break; + + case 0x80: + { + int samples, bytes; + int64_t ticks; + + if (-1 == s->freq) + s->freq = 11025; + samples = dsp_get_lohi (s); + bytes = samples << s->fmt_stereo << (s->fmt_bits == 16); + ticks = ticks_per_sec / (s->freq / bytes); + if (ticks < ticks_per_sec / 1024) + pic_set_irq (s->irq, 1); + else + qemu_mod_timer (s->aux_ts, qemu_get_clock (vm_clock) + ticks); + ldebug ("mix silence %d %d %lld\n", samples, bytes, ticks); + } break; case 0xe0: - d0 = dsp_get_data (dsp); - dsp->out_data_len = 0; - linfo ("data = %#x\n", d0); - dsp_out_data (dsp, d0 ^ 0xff); + d0 = dsp_get_data (s); + s->out_data_len = 0; + ldebug ("E0 data = %#x\n", d0); + dsp_out_data(s, ~d0); break; - case 0xe4: - dsp->test_reg = dsp_get_data (dsp); + case 0xe2: + d0 = dsp_get_data (s); + dolog ("E2 = %#x\n", d0); break; + case 0xe4: + s->test_reg = dsp_get_data (s); + break; case 0xf9: - d0 = dsp_get_data (dsp); - ldebug ("f9 <- %#x\n", d0); + d0 = dsp_get_data (s); + ldebug ("command 0xf9 with %#x\n", d0); switch (d0) { - case 0x0e: dsp_out_data (dsp, 0xff); break; - case 0x0f: dsp_out_data (dsp, 0x07); break; - case 0xf9: dsp_out_data (dsp, 0x00); break; + case 0x0e: + dsp_out_data (s, 0xff); + break; + + case 0x0f: + dsp_out_data (s, 0x07); + break; + case 0x37: - dsp_out_data (dsp, 0x38); break; + dsp_out_data (s, 0x38); + break; + default: - dsp_out_data (dsp, 0); + dsp_out_data (s, 0x00); + break; } break; default: - dolog ("complete: unrecognized command %#x\n", dsp->cmd); + dolog ("complete: unrecognized command %#x\n", s->cmd); return; } } - dsp->needed_bytes = 0; - dsp->cmd = -1; + ldebug ("\n"); + s->cmd = -1; return; } +static void reset (SB16State *s) +{ + pic_set_irq (s->irq, 0); + if (s->dma_auto) { + pic_set_irq (s->irq, 1); + pic_set_irq (s->irq, 0); + } + + s->mixer_regs[0x82] = 0; + s->dma_auto = 0; + s->in_index = 0; + s->out_data_len = 0; + s->left_till_irq = 0; + s->needed_bytes = 0; + s->block_size = -1; + s->nzero = 0; + s->highspeed = 0; + s->v2x6 = 0; + + dsp_out_data(s, 0xaa); + speaker (s, 0); + control (s, 0); +} + static IO_WRITE_PROTO (dsp_write) { - SB16State *dsp = opaque; + SB16State *s = opaque; int iport; - iport = nport - sb.port; + iport = nport - s->port; - ldebug ("dsp_write %#x <- %#x\n", nport, val); + ldebug ("write %#x <- %#x\n", nport, val); switch (iport) { - case 0x6: - control (0); - if (0 == val) - dsp->v2x6 = 0; - else if ((1 == val) && (0 == dsp->v2x6)) { - dsp->v2x6 = 1; - dsp->dma_pos = 0; - dsp->dma_auto = 0; - dsp->in_index = 0; - dsp->out_data_len = 0; - dsp->left_till_irq = 0; - dsp->speaker = 0; - dsp->needed_bytes = 0; - pic_set_irq (sb.irq, 0); - dsp_out_data(dsp, 0xaa); + case 0x06: + switch (val) { + case 0x00: + if (s->v2x6 == 1) { + if (0 && s->highspeed) { + s->highspeed = 0; + pic_set_irq (s->irq, 0); + control (s, 0); + } + else + reset (s); + } + s->v2x6 = 0; + break; + + case 0x01: + case 0x03: /* FreeBSD kludge */ + s->v2x6 = 1; + break; + + case 0xc6: + s->v2x6 = 0; /* Prince of Persia, csp.sys, diagnose.exe */ + break; + + case 0xb8: /* Panic */ + reset (s); + break; + + case 0x39: + dsp_out_data (s, 0x38); + reset (s); + s->v2x6 = 0x39; + break; + + default: + s->v2x6 = val; + break; } - else - dsp->v2x6 = ~0; break; - case 0xc: /* write data or command | write status */ - if (0 == dsp->needed_bytes) { - command (dsp, val); - if (0 == dsp->needed_bytes) { - log_dsp (dsp); + case 0x0c: /* write data or command | write status */ +/* if (s->highspeed) */ +/* break; */ + + if (0 == s->needed_bytes) { + command (s, val); +#if 0 + if (0 == s->needed_bytes) { + log_dsp (s); } +#endif } else { - if (dsp->in_index == sizeof (dsp->in2_data)) { + if (s->in_index == sizeof (s->in2_data)) { dolog ("in data overrun\n"); } else { - dsp->in2_data[dsp->in_index++] = val; - if (dsp->in_index == dsp->needed_bytes) { - dsp->needed_bytes = 0; - complete (dsp); - log_dsp (dsp); + s->in2_data[s->in_index++] = val; + if (s->in_index == s->needed_bytes) { + s->needed_bytes = 0; + complete (s); +#if 0 + log_dsp (s); +#endif + } } } - } break; default: - dolog ("dsp_write (nport=%#x, val=%#x)\n", nport, val); + ldebug ("(nport=%#x, val=%#x)\n", nport, val); break; } } static IO_READ_PROTO (dsp_read) { - SB16State *dsp = opaque; - int iport, retval; + SB16State *s = opaque; + int iport, retval, ack = 0; - iport = nport - sb.port; + iport = nport - s->port; switch (iport) { - case 0x6: /* reset */ - control (0); - retval = 0; - dsp->speaker = 0; + case 0x06: /* reset */ + retval = 0xff; break; - case 0xa: /* read data */ - if (dsp->out_data_len) { - retval = dsp->out_data[--dsp->out_data_len]; - dsp->last_read_byte = retval; - } else { - retval = dsp->last_read_byte; + case 0x0a: /* read data */ + if (s->out_data_len) { + retval = s->out_data[--s->out_data_len]; + s->last_read_byte = retval; + } + else { dolog ("empty output buffer\n"); + retval = s->last_read_byte; /* goto error; */ } break; - case 0xc: /* 0xxxxxxx can write */ - retval = 0; - if (dsp->out_data_len == sizeof (dsp->out_data)) retval |= 0x80; + case 0x0c: /* 0 can write */ + retval = s->can_write ? 0 : 0x80; break; - case 0xd: /* timer interrupt clear */ - dolog ("timer interrupt clear\n"); - goto error; + case 0x0d: /* timer interrupt clear */ + /* dolog ("timer interrupt clear\n"); */ + retval = 0; + break; - case 0xe: /* data available status | irq 8 ack */ - /* XXX drop pic irq line here? */ - /* ldebug ("8 ack\n"); */ - retval = dsp->out_data_len ? 0x80 : 0; - dsp->mixer_regs[0x82] &= ~dsp->mixer_regs[0x80]; - pic_set_irq (sb.irq, 0); + case 0x0e: /* data available status | irq 8 ack */ + retval = (!s->out_data_len || s->highspeed) ? 0 : 0x80; + if (s->mixer_regs[0x82] & 1) { + ack = 1; + s->mixer_regs[0x82] &= 1; + pic_set_irq (s->irq, 0); + } break; - case 0xf: /* irq 16 ack */ - /* XXX drop pic irq line here? */ - ldebug ("16 ack\n"); + case 0x0f: /* irq 16 ack */ retval = 0xff; - dsp->mixer_regs[0x82] &= ~dsp->mixer_regs[0x80]; - pic_set_irq (sb.irq, 0); + if (s->mixer_regs[0x82] & 2) { + ack = 1; + s->mixer_regs[0x82] &= 2; + pic_set_irq (s->irq, 0); + } break; default: goto error; } - if (0xe == iport) { - if (0 == retval) { - if (!dsp->nzero) { - ldebug ("dsp_read (nport=%#x iport %#x) = %#x, %lld\n", - nport, iport, retval, dsp->nzero); - } - dsp->nzero += 1; - } - else { - ldebug ("dsp_read (nport=%#x iport %#x) = %#x, %lld\n", - nport, iport, retval, dsp->nzero); - dsp->nzero = 0; - } - } - else { - ldebug ("dsp_read (nport=%#x iport %#x) = %#x\n", - nport, iport, retval); - } + if (!ack) + ldebug ("read %#x -> %#x\n", nport, retval); return retval; error: - printf ("dsp_read error %#x\n", nport); + dolog ("WARNING dsp_read %#x error\n", nport); return 0xff; } +static void reset_mixer (SB16State *s) +{ + int i; + + memset (s->mixer_regs, 0xff, 0x7f); + memset (s->mixer_regs + 0x83, 0xff, sizeof (s->mixer_regs) - 0x83); + + s->mixer_regs[0x02] = 4; /* master volume 3bits */ + s->mixer_regs[0x06] = 4; /* MIDI volume 3bits */ + s->mixer_regs[0x08] = 0; /* CD volume 3bits */ + s->mixer_regs[0x0a] = 0; /* voice volume 2bits */ + + /* d5=input filt, d3=lowpass filt, d1,d2=input source */ + s->mixer_regs[0x0c] = 0; + + /* d5=output filt, d1=stereo switch */ + s->mixer_regs[0x0e] = 0; + + /* voice volume L d5,d7, R d1,d3 */ + s->mixer_regs[0x04] = (4 << 5) | (4 << 1); + /* master ... */ + s->mixer_regs[0x22] = (4 << 5) | (4 << 1); + /* MIDI ... */ + s->mixer_regs[0x26] = (4 << 5) | (4 << 1); + + for (i = 0x30; i < 0x48; i++) { + s->mixer_regs[i] = 0x20; + } +} + static IO_WRITE_PROTO(mixer_write_indexb) { - SB16State *dsp = opaque; - dsp->mixer_nreg = val; + SB16State *s = opaque; + s->mixer_nreg = val; } static IO_WRITE_PROTO(mixer_write_datab) { - SB16State *dsp = opaque; - int i; + SB16State *s = opaque; + + ldebug ("mixer_write [%#x] <- %#x\n", s->mixer_nreg, val); + if (s->mixer_nreg > sizeof (s->mixer_regs)) + return; - linfo ("mixer [%#x] <- %#x\n", dsp->mixer_nreg, val); - switch (dsp->mixer_nreg) { + switch (s->mixer_nreg) { case 0x00: - /* Bochs */ - dsp->mixer_regs[0x04] = 0xcc; - dsp->mixer_regs[0x0a] = 0x00; - dsp->mixer_regs[0x22] = 0xcc; - dsp->mixer_regs[0x26] = 0xcc; - dsp->mixer_regs[0x28] = 0x00; - dsp->mixer_regs[0x2e] = 0x00; - dsp->mixer_regs[0x3c] = 0x1f; - dsp->mixer_regs[0x3d] = 0x15; - dsp->mixer_regs[0x3e] = 0x0b; - - for (i = 0x30; i <= 0x35; i++) - dsp->mixer_regs[i] = 0xc0; - - for (i = 0x36; i <= 0x3b; i++) - dsp->mixer_regs[i] = 0x00; - - for (i = 0x3f; i <= 0x43; i++) - dsp->mixer_regs[i] = 0x00; - - for (i = 0x44; i <= 0x47; i++) - dsp->mixer_regs[i] = 0x80; - - for (i = 0x30; i < 0x48; i++) { - dsp->mixer_regs[i] = 0x20; - } + reset_mixer (s); break; - case 0x04: - case 0x0a: - case 0x22: - case 0x26: - case 0x28: - case 0x2e: - case 0x30: - case 0x31: - case 0x32: - case 0x33: - case 0x34: - case 0x35: - case 0x36: - case 0x37: - case 0x38: - case 0x39: - case 0x3a: - case 0x3b: - case 0x3c: - case 0x3d: - case 0x3e: - case 0x3f: - case 0x40: - case 0x41: - case 0x42: - case 0x43: - case 0x44: - case 0x45: - case 0x46: - case 0x47: case 0x80: - case 0x81: + { + int irq = irq_of_magic (val); + ldebug ("setting irq to %d (val=%#x)\n", irq, val); + if (irq > 0) + s->irq = irq; + } break; - default: - return; - } - dsp->mixer_regs[dsp->mixer_nreg] = val; -} -static IO_WRITE_PROTO(mpu_write) -{ - linfo ("mpu: %#x\n", val); -} + case 0x81: + { + int dma, hdma; -static IO_WRITE_PROTO(adlib_write) -{ - linfo ("adlib: %#x\n", val); -} + dma = lsbindex (val & 0xf); + hdma = lsbindex (val & 0xf0); + dolog ("attempt to set DMA register 8bit %d, 16bit %d (val=%#x)\n", + dma, hdma, val); +#if 0 + s->dma = dma; + s->hdma = hdma; +#endif + } + break; -static IO_READ_PROTO(mpu_read) -{ - linfo ("mpu read: %#x\n", nport); - return 0x80; -} + case 0x82: + dolog ("attempt to write into IRQ status register (val=%#x)\n", + val); + return; -static IO_READ_PROTO(adlib_read) -{ - linfo ("adlib read: %#x\n", nport); - return 0; + default: + if (s->mixer_nreg >= 0x80) + dolog ("attempt to write mixer[%#x] <- %#x\n", s->mixer_nreg, val); + break; + } + + s->mixer_regs[s->mixer_nreg] = val; } static IO_WRITE_PROTO(mixer_write_indexw) @@ -765,194 +962,229 @@ static IO_WRITE_PROTO(mixer_write_indexw) static IO_READ_PROTO(mixer_read) { - SB16State *dsp = opaque; - linfo ("mixer [%#x] -> %#x\n", dsp->mixer_nreg, dsp->mixer_regs[dsp->mixer_nreg]); - return dsp->mixer_regs[dsp->mixer_nreg]; -} - -void SB16_run (void) -{ - if (0 == dsp.speaker) - return; - - AUD_run (); + SB16State *s = opaque; + ldebug ("mixer_read[%#x] -> %#x\n", + s->mixer_nreg, s->mixer_regs[s->mixer_nreg]); + return s->mixer_regs[s->mixer_nreg]; } -static int write_audio (uint32_t addr, int len, int size) +static int write_audio (SB16State *s, int nchan, int dma_pos, + int dma_len, int len) { int temp, net; uint8_t tmpbuf[4096]; - temp = size; - + temp = len; net = 0; while (temp) { - int left_till_end; - int to_copy; - int copied; + int left = dma_len - dma_pos; + int to_copy, copied; - left_till_end = len - dsp.dma_pos; - - to_copy = MIN (temp, left_till_end); + to_copy = audio_MIN (temp, left); if (to_copy > sizeof(tmpbuf)) to_copy = sizeof(tmpbuf); - cpu_physical_memory_read(addr + dsp.dma_pos, tmpbuf, to_copy); - copied = AUD_write (tmpbuf, to_copy); - temp -= copied; - dsp.dma_pos += copied; - - if (dsp.dma_pos == len) { - dsp.dma_pos = 0; - } + copied = DMA_read_memory (nchan, tmpbuf, dma_pos, to_copy); + copied = AUD_write (s->voice, tmpbuf, copied); + temp -= copied; + dma_pos = (dma_pos + copied) % dma_len; net += copied; - if (copied != to_copy) - return net; + if (!copied) + break; } return net; } -static int SB_read_DMA (void *opaque, target_ulong addr, int size) +static int SB_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len) { - SB16State *dsp = opaque; - int free, till, copy, written; - - if (0 == dsp->speaker) - return 0; + SB16State *s = opaque; + int free, rfree, till, copy, written, elapsed; - if (dsp->left_till_irq < 0) { - ldebug ("left_till_irq < 0, %d, pos %d \n", - dsp->left_till_irq, dsp->dma_buffer_size); - dsp->left_till_irq += dsp->dma_buffer_size; - return dsp->dma_pos; + if (s->left_till_irq < 0) { + s->left_till_irq = s->block_size; } - free = AUD_get_free (); - - if ((free <= 0) || (0 == size)) { - ldebug ("returning, since free = %d and size = %d\n", free, size); - return dsp->dma_pos; - } + elapsed = AUD_calc_elapsed (s->voice); + free = elapsed;/* AUD_get_free (s->voice); */ + rfree = free; + free = audio_MIN (free, elapsed) & ~s->align; - if (mix_block > 0) { - copy = MIN (free, mix_block); - } - else { - copy = free; + if ((free <= 0) || !dma_len) { + return dma_pos; } - till = dsp->left_till_irq; + copy = free; + till = s->left_till_irq; #ifdef DEBUG_SB16_MOST - ldebug ("addr:%#010x free:%d till:%d size:%d\n", - addr, free, till, size); + dolog ("pos:%06d free:%d,%d till:%d len:%d\n", + dma_pos, free, AUD_get_free (s->voice), till, dma_len); #endif if (till <= copy) { - if (0 == dsp->dma_auto) { + if (0 == s->dma_auto) { copy = till; } } - written = write_audio (addr, size, copy); - dsp->left_till_irq -= written; - AUD_adjust_estimate (free - written); + written = write_audio (s, nchan, dma_pos, dma_len, copy); + dma_pos = (dma_pos + written) % dma_len; + s->left_till_irq -= written; - if (dsp->left_till_irq <= 0) { - dsp->mixer_regs[0x82] |= dsp->mixer_regs[0x80]; - if (0 == noirq) { - ldebug ("request irq pos %d, left %d\n", - dsp->dma_pos, dsp->left_till_irq); - pic_set_irq(sb.irq, 1); - } - - if (0 == dsp->dma_auto) { - control (0); + if (s->left_till_irq <= 0) { + s->mixer_regs[0x82] |= (nchan & 4) ? 2 : 1; + pic_set_irq (s->irq, 1); + if (0 == s->dma_auto) { + control (s, 0); + speaker (s, 0); } } #ifdef DEBUG_SB16_MOST ldebug ("pos %5d free %5d size %5d till % 5d copy %5d dma size %5d\n", - dsp->dma_pos, free, size, dsp->left_till_irq, copy, - dsp->dma_buffer_size); + dma_pos, free, dma_len, s->left_till_irq, copy, s->block_size); #endif - if (dsp->left_till_irq <= 0) { - dsp->left_till_irq += dsp->dma_buffer_size; + while (s->left_till_irq <= 0) { + s->left_till_irq = s->block_size + s->left_till_irq; } - return dsp->dma_pos; -} - -static int magic_of_irq (int irq) -{ - switch (irq) { - case 2: - return 1; - case 5: - return 2; - case 7: - return 4; - case 10: - return 8; - default: - dolog ("bad irq %d\n", irq); - return 2; - } + AUD_adjust (s->voice, written); + return dma_pos; } -#if 0 -static int irq_of_magic (int magic) +void SB_timer (void *opaque) { - switch (magic) { - case 1: - return 2; - case 2: - return 5; - case 4: - return 7; - case 8: - return 10; - default: - dolog ("bad irq magic %d\n", magic); - return 2; - } + SB16State *s = opaque; + AUD_run (); + qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1); } -#endif -#ifdef SB16_TRAP_ALL -static IO_READ_PROTO (trap_read) +static void SB_save (QEMUFile *f, void *opaque) { - switch (nport) { - case 0x220: - return 0; - case 0x226: - case 0x22a: - case 0x22c: - case 0x22d: - case 0x22e: - case 0x22f: - return dsp_read (opaque, nport); - } - linfo ("trap_read: %#x\n", nport); - return 0xff; + SB16State *s = opaque; + + qemu_put_be32s (f, &s->irq); + qemu_put_be32s (f, &s->dma); + qemu_put_be32s (f, &s->hdma); + qemu_put_be32s (f, &s->port); + qemu_put_be32s (f, &s->ver); + qemu_put_be32s (f, &s->in_index); + qemu_put_be32s (f, &s->out_data_len); + qemu_put_be32s (f, &s->fmt_stereo); + qemu_put_be32s (f, &s->fmt_signed); + qemu_put_be32s (f, &s->fmt_bits); + qemu_put_be32s (f, &s->fmt); + qemu_put_be32s (f, &s->dma_auto); + qemu_put_be32s (f, &s->block_size); + qemu_put_be32s (f, &s->fifo); + qemu_put_be32s (f, &s->freq); + qemu_put_be32s (f, &s->time_const); + qemu_put_be32s (f, &s->speaker); + qemu_put_be32s (f, &s->needed_bytes); + qemu_put_be32s (f, &s->cmd); + qemu_put_be32s (f, &s->use_hdma); + qemu_put_be32s (f, &s->highspeed); + qemu_put_be32s (f, &s->can_write); + qemu_put_be32s (f, &s->v2x6); + + qemu_put_8s (f, &s->csp_param); + qemu_put_8s (f, &s->csp_value); + qemu_put_8s (f, &s->csp_mode); + qemu_put_8s (f, &s->csp_param); + qemu_put_buffer (f, s->csp_regs, 256); + qemu_put_8s (f, &s->csp_index); + qemu_put_buffer (f, s->csp_reg83, 4); + qemu_put_be32s (f, &s->csp_reg83r); + qemu_put_be32s (f, &s->csp_reg83w); + + qemu_put_buffer (f, s->in2_data, sizeof (s->in2_data)); + qemu_put_buffer (f, s->out_data, sizeof (s->out_data)); + qemu_put_8s (f, &s->test_reg); + qemu_put_8s (f, &s->last_read_byte); + + qemu_put_be32s (f, &s->nzero); + qemu_put_be32s (f, &s->left_till_irq); + qemu_put_be32s (f, &s->dma_running); + qemu_put_be32s (f, &s->bytes_per_second); + qemu_put_be32s (f, &s->align); + + qemu_put_be32s (f, &s->mixer_nreg); + qemu_put_buffer (f, s->mixer_regs, 256); } -static IO_WRITE_PROTO (trap_write) +static int SB_load (QEMUFile *f, void *opaque, int version_id) { - switch (nport) { - case 0x226: - case 0x22c: - dsp_write (opaque, nport, val); - return; + SB16State *s = opaque; + + if (version_id != 1) + return -EINVAL; + + qemu_get_be32s (f, &s->irq); + qemu_get_be32s (f, &s->dma); + qemu_get_be32s (f, &s->hdma); + qemu_get_be32s (f, &s->port); + qemu_get_be32s (f, &s->ver); + qemu_get_be32s (f, &s->in_index); + qemu_get_be32s (f, &s->out_data_len); + qemu_get_be32s (f, &s->fmt_stereo); + qemu_get_be32s (f, &s->fmt_signed); + qemu_get_be32s (f, &s->fmt_bits); + qemu_get_be32s (f, &s->fmt); + qemu_get_be32s (f, &s->dma_auto); + qemu_get_be32s (f, &s->block_size); + qemu_get_be32s (f, &s->fifo); + qemu_get_be32s (f, &s->freq); + qemu_get_be32s (f, &s->time_const); + qemu_get_be32s (f, &s->speaker); + qemu_get_be32s (f, &s->needed_bytes); + qemu_get_be32s (f, &s->cmd); + qemu_get_be32s (f, &s->use_hdma); + qemu_get_be32s (f, &s->highspeed); + qemu_get_be32s (f, &s->can_write); + qemu_get_be32s (f, &s->v2x6); + + qemu_get_8s (f, &s->csp_param); + qemu_get_8s (f, &s->csp_value); + qemu_get_8s (f, &s->csp_mode); + qemu_get_8s (f, &s->csp_param); + qemu_get_buffer (f, s->csp_regs, 256); + qemu_get_8s (f, &s->csp_index); + qemu_get_buffer (f, s->csp_reg83, 4); + qemu_get_be32s (f, &s->csp_reg83r); + qemu_get_be32s (f, &s->csp_reg83w); + + qemu_get_buffer (f, s->in2_data, sizeof (s->in2_data)); + qemu_get_buffer (f, s->out_data, sizeof (s->out_data)); + qemu_get_8s (f, &s->test_reg); + qemu_get_8s (f, &s->last_read_byte); + + qemu_get_be32s (f, &s->nzero); + qemu_get_be32s (f, &s->left_till_irq); + qemu_get_be32s (f, &s->dma_running); + qemu_get_be32s (f, &s->bytes_per_second); + qemu_get_be32s (f, &s->align); + + qemu_get_be32s (f, &s->mixer_nreg); + qemu_get_buffer (f, s->mixer_regs, 256); + + if (s->voice) + AUD_reset (s->voice); + + if (s->dma_running) { + if (s->freq) + s->voice = AUD_open (s->voice, "sb16", s->freq, + 1 << s->fmt_stereo, s->fmt); + + control (s, 1); + speaker (s, s->speaker); } - linfo ("trap_write: %#x = %#x\n", nport, val); + return 0; } -#endif void SB16_init (void) { @@ -961,47 +1193,45 @@ void SB16_init (void) static const uint8_t dsp_write_ports[] = {0x6, 0xc}; static const uint8_t dsp_read_ports[] = {0x6, 0xa, 0xc, 0xd, 0xe, 0xf}; - memset(s->mixer_regs, 0xff, sizeof(s->mixer_regs)); + s->ts = qemu_new_timer (vm_clock, SB_timer, s); + if (!s->ts) + return; + + s->irq = conf.irq; + s->dma = conf.dma; + s->hdma = conf.hdma; + s->port = conf.port; + s->ver = conf.ver_lo | (conf.ver_hi << 8); - s->mixer_regs[0x00] = 0; - s->mixer_regs[0x0e] = ~0; - s->mixer_regs[0x80] = magic_of_irq (sb.irq); - s->mixer_regs[0x81] = 0x80 | 0x10 | (sb.dma << 1); - s->mixer_regs[0x82] = 0; - s->mixer_regs[0xfd] = 16; /* bochs */ - s->mixer_regs[0xfe] = 6; /* bochs */ - mixer_write_indexw (s, 0x224, 0); - -#ifdef SB16_TRAP_ALL - for (i = 0; i < 0x100; i++) { - if (i != 4 && i != 5) { - register_ioport_write (sb.port + i, 1, 1, trap_write, s); - register_ioport_read (sb.port + i, 1, 1, trap_read, s); - } - } -#else + s->mixer_regs[0x80] = magic_of_irq (s->irq); + s->mixer_regs[0x81] = (1 << s->dma) | (1 << s->hdma); + s->mixer_regs[0x82] = 2 << 5; + + s->csp_regs[5] = 1; + s->csp_regs[9] = 0xf8; + + reset_mixer (s); + s->aux_ts = qemu_new_timer (vm_clock, aux_timer, s); + if (!s->aux_ts) + return; for (i = 0; i < LENOFA (dsp_write_ports); i++) { - register_ioport_write (sb.port + dsp_write_ports[i], 1, 1, dsp_write, s); + register_ioport_write (s->port + dsp_write_ports[i], 1, 1, dsp_write, s); } for (i = 0; i < LENOFA (dsp_read_ports); i++) { - register_ioport_read (sb.port + dsp_read_ports[i], 1, 1, dsp_read, s); + register_ioport_read (s->port + dsp_read_ports[i], 1, 1, dsp_read, s); } -#endif - register_ioport_write (sb.port + 0x4, 1, 1, mixer_write_indexb, s); - register_ioport_write (sb.port + 0x4, 1, 2, mixer_write_indexw, s); - register_ioport_read (sb.port + 0x5, 1, 1, mixer_read, s); - register_ioport_write (sb.port + 0x5, 1, 1, mixer_write_datab, s); + register_ioport_write (s->port + 0x4, 1, 1, mixer_write_indexb, s); + register_ioport_write (s->port + 0x4, 1, 2, mixer_write_indexw, s); + register_ioport_read (s->port + 0x5, 1, 1, mixer_read, s); + register_ioport_write (s->port + 0x5, 1, 1, mixer_write_datab, s); - for (i = 0; 4 < 4; i++) { - register_ioport_read (0x330 + i, 1, 1, mpu_read, s); - register_ioport_write (0x330 + i, 1, 1, mpu_write, s); - register_ioport_read (0x388 + i, 1, 1, adlib_read, s); - register_ioport_write (0x388 + i, 1, 1, adlib_write, s); - } + DMA_register_channel (s->hdma, SB_read_DMA, s); + DMA_register_channel (s->dma, SB_read_DMA, s); + s->can_write = 1; - DMA_register_channel (sb.hdma, SB_read_DMA, s); - DMA_register_channel (sb.dma, SB_read_DMA, s); + qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1); + register_savevm ("sb16", 0, 1, SB_save, SB_load, s); } diff --git a/oss.c b/oss.c deleted file mode 100644 index 91eb47e491..0000000000 --- a/oss.c +++ /dev/null @@ -1,978 +0,0 @@ -/* - * QEMU OSS Audio output driver - * - * Copyright (c) 2003 Vassili Karpov (malc) - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ -#include "vl.h" - -#include <stdio.h> -#include <limits.h> -#include <stdlib.h> -#include <string.h> - -/* TODO: Graceful error handling */ - -#if defined(_WIN32) -#define USE_SDL_AUDIO -#endif - -#define MIN(a, b) ((a)>(b)?(b):(a)) -#define MAX(a, b) ((a)<(b)?(b):(a)) - -#define DEREF(x) (void)x -#define dolog(...) fprintf (stderr, "audio: " __VA_ARGS__) -#define ERRFail(...) do { \ - int _errno = errno; \ - fprintf (stderr, "audio: " __VA_ARGS__); \ - fprintf (stderr, "\nsystem error: %s\n", strerror (_errno)); \ - abort (); \ -} while (0) -#define Fail(...) do { \ - fprintf (stderr, "audio: " __VA_ARGS__); \ - fprintf (stderr, "\n"); \ - abort (); \ -} while (0) - -#ifdef DEBUG_AUDIO -#define lwarn(...) fprintf (stderr, "audio: " __VA_ARGS__) -#define linfo(...) fprintf (stderr, "audio: " __VA_ARGS__) -#define ldebug(...) fprintf (stderr, "audio: " __VA_ARGS__) -#else -#define lwarn(...) -#define linfo(...) -#define ldebug(...) -#endif - -static int get_conf_val (const char *key, int defval) -{ - int val = defval; - char *strval; - - strval = getenv (key); - if (strval) { - val = atoi (strval); - } - - return val; -} - -static void copy_no_conversion (void *dst, void *src, int size) -{ - memcpy (dst, src, size); -} - -static void copy_u16_to_s16 (void *dst, void *src, int size) -{ - int i; - uint16_t *out, *in; - - out = dst; - in = src; - - for (i = 0; i < size / 2; i++) { - out[i] = in[i] + 0x8000; - } -} - -#ifdef USE_SDL_AUDIO -#include <SDL/SDL.h> -#include <SDL/SDL_thread.h> - -static struct { - int samples; -} conf = { - .samples = 4096 -}; - -typedef struct AudioState { - int freq; - int bits16; - int nchannels; - int rpos; - int wpos; - volatile int live; - volatile int exit; - int bytes_per_second; - Uint8 *buf; - int bufsize; - int leftover; - uint64_t old_ticks; - SDL_AudioSpec spec; - SDL_mutex *mutex; - SDL_sem *sem; - void (*copy_fn)(void *, void *, int); -} AudioState; - -static AudioState sdl_audio; - -void AUD_run (void) -{ -} - -static void own (AudioState *s) -{ - /* SDL_LockAudio (); */ - if (SDL_mutexP (s->mutex)) - dolog ("SDL_mutexP: %s\n", SDL_GetError ()); -} - -static void disown (AudioState *s) -{ - /* SDL_UnlockAudio (); */ - if (SDL_mutexV (s->mutex)) - dolog ("SDL_mutexV: %s\n", SDL_GetError ()); -} - -static void sem_wait (AudioState *s) -{ - if (SDL_SemWait (s->sem)) - dolog ("SDL_SemWait: %s\n", SDL_GetError ()); -} - -static void sem_post (AudioState *s) -{ - if (SDL_SemPost (s->sem)) - dolog ("SDL_SemPost: %s\n", SDL_GetError ()); -} - -static void audio_callback (void *data, Uint8 *stream, int len) -{ - int to_mix; - AudioState *s = data; - - if (s->exit) return; - while (len) { - sem_wait (s); - if (s->exit) return; - own (s); - to_mix = MIN (len, s->live); - len -= to_mix; - /* printf ("to_mix=%d len=%d live=%d\n", to_mix, len, s->live); */ - while (to_mix) { - int chunk = MIN (to_mix, s->bufsize - s->rpos); - /* SDL_MixAudio (stream, buf, chunk, SDL_MIX_MAXVOLUME); */ - memcpy (stream, s->buf + s->rpos, chunk); - - s->rpos += chunk; - s->live -= chunk; - - stream += chunk; - to_mix -= chunk; - - if (s->rpos == s->bufsize) s->rpos = 0; - } - disown (s); - } -} - -static void sem_zero (AudioState *s) -{ - int res; - - do { - res = SDL_SemTryWait (s->sem); - if (res < 0) { - dolog ("SDL_SemTryWait: %s\n", SDL_GetError ()); - return; - } - } while (res != SDL_MUTEX_TIMEDOUT); -} - -static void do_open (AudioState *s) -{ - int status; - SDL_AudioSpec obtained; - - SDL_PauseAudio (1); - if (s->buf) { - s->exit = 1; - sem_post (s); - SDL_CloseAudio (); - s->exit = 0; - qemu_free (s->buf); - s->buf = NULL; - sem_zero (s); - } - - s->bytes_per_second = (s->spec.freq << (s->spec.channels >> 1)) << s->bits16; - s->spec.samples = conf.samples; - s->spec.userdata = s; - s->spec.callback = audio_callback; - - status = SDL_OpenAudio (&s->spec, &obtained); - if (status < 0) { - dolog ("SDL_OpenAudio: %s\n", SDL_GetError ()); - goto exit; - } - - if (obtained.freq != s->spec.freq || - obtained.channels != s->spec.channels || - obtained.format != s->spec.format) { - dolog ("Audio spec mismatch requested obtained\n" - "freq %5d %5d\n" - "channels %5d %5d\n" - "fmt %5d %5d\n", - s->spec.freq, obtained.freq, - s->spec.channels, obtained.channels, - s->spec.format, obtained.format - ); - } - - s->bufsize = obtained.size; - s->buf = qemu_mallocz (s->bufsize); - if (!s->buf) { - dolog ("qemu_mallocz(%d)\n", s->bufsize); - goto exit; - } - SDL_PauseAudio (0); - -exit: - s->rpos = 0; - s->wpos = 0; - s->live = 0; -} - -int AUD_write (void *in_buf, int size) -{ - AudioState *s = &sdl_audio; - int to_copy, temp; - uint8_t *in, *out; - - own (s); - to_copy = MIN (s->bufsize - s->live, size); - - temp = to_copy; - - in = in_buf; - out = s->buf; - - while (temp) { - int copy; - - copy = MIN (temp, s->bufsize - s->wpos); - s->copy_fn (out + s->wpos, in, copy); - - s->wpos += copy; - if (s->wpos == s->bufsize) { - s->wpos = 0; - } - - temp -= copy; - in += copy; - s->live += copy; - } - - disown (s); - sem_post (s); - return to_copy; -} - -static void maybe_open (AudioState *s, int req_freq, int req_nchannels, - audfmt_e req_fmt, int force_open) -{ - int sdl_fmt, bits16; - - switch (req_fmt) { - case AUD_FMT_U8: - bits16 = 0; - sdl_fmt = AUDIO_U8; - s->copy_fn = copy_no_conversion; - break; - - case AUD_FMT_S8: - fprintf (stderr, "audio: can not play 8bit signed\n"); - return; - - case AUD_FMT_S16: - bits16 = 1; - sdl_fmt = AUDIO_S16; - s->copy_fn = copy_no_conversion; - break; - - case AUD_FMT_U16: - bits16 = 1; - sdl_fmt = AUDIO_S16; - s->copy_fn = copy_u16_to_s16; - break; - - default: - abort (); - } - - if (force_open - || (NULL == s->buf) - || (sdl_fmt != s->spec.format) - || (req_nchannels != s->spec.channels) - || (req_freq != s->spec.freq) - || (bits16 != s->bits16)) { - - s->spec.format = sdl_fmt; - s->spec.channels = req_nchannels; - s->spec.freq = req_freq; - s->bits16 = bits16; - do_open (s); - } -} - -void AUD_reset (int req_freq, int req_nchannels, audfmt_e req_fmt) -{ - AudioState *s = &sdl_audio; - own (s); - maybe_open (s, req_freq, req_nchannels, req_fmt, 0); - disown (s); -} - -void AUD_open (int req_freq, int req_nchannels, audfmt_e req_fmt) -{ - AudioState *s = &sdl_audio; - own (s); - maybe_open (s, req_freq, req_nchannels, req_fmt, 1); - disown (s); -} - -void AUD_adjust_estimate (int leftover) -{ - AudioState *s = &sdl_audio; - own (s); - s->leftover = leftover; - disown (s); -} - -int AUD_get_free (void) -{ - int free, elapsed; - uint64_t ticks, delta; - uint64_t ua_elapsed; - uint64_t al_elapsed; - AudioState *s = &sdl_audio; - - own (s); - free = s->bufsize - s->live; - - if (0 == free) { - disown (s); - return 0; - } - - elapsed = free; - ticks = qemu_get_clock(rt_clock); - delta = ticks - s->old_ticks; - s->old_ticks = ticks; - - ua_elapsed = (delta * s->bytes_per_second) / 1000; - al_elapsed = ua_elapsed & ~3ULL; - - ldebug ("tid elapsed %llu bytes\n", ua_elapsed); - - if (al_elapsed > (uint64_t) INT_MAX) - elapsed = INT_MAX; - else - elapsed = al_elapsed; - - elapsed += s->leftover; - disown (s); - - if (elapsed > free) { - lwarn ("audio can not keep up elapsed %d free %d\n", elapsed, free); - return free; - } - else { - return elapsed; - } -} - -int AUD_get_live (void) -{ - int live; - AudioState *s = &sdl_audio; - - own (s); - live = s->live; - disown (s); - return live; -} - -int AUD_get_buffer_size (void) -{ - int bufsize; - AudioState *s = &sdl_audio; - - own (s); - bufsize = s->bufsize; - disown (s); - return bufsize; -} - -#define QC_SDL_NSAMPLES "QEMU_SDL_NSAMPLES" - -static void cleanup (void) -{ - AudioState *s = &sdl_audio; - own (s); - s->exit = 1; - sem_post (s); - disown (s); -} - -void AUD_init (void) -{ - AudioState *s = &sdl_audio; - - atexit (cleanup); - SDL_InitSubSystem (SDL_INIT_AUDIO); - s->mutex = SDL_CreateMutex (); - if (!s->mutex) { - dolog ("SDL_CreateMutex: %s\n", SDL_GetError ()); - return; - } - - s->sem = SDL_CreateSemaphore (0); - if (!s->sem) { - dolog ("SDL_CreateSemaphore: %s\n", SDL_GetError ()); - return; - } - - conf.samples = get_conf_val (QC_SDL_NSAMPLES, conf.samples); -} - -#elif !defined(_WIN32) && !defined(__APPLE__) - -#include <fcntl.h> -#include <errno.h> -#include <unistd.h> -#include <sys/mman.h> -#include <sys/types.h> -#include <sys/ioctl.h> -#include <sys/soundcard.h> - -/* http://www.df.lth.se/~john_e/gems/gem002d.html */ -/* http://www.multi-platforms.com/Tips/PopCount.htm */ -static inline uint32_t popcount (uint32_t u) -{ - u = ((u&0x55555555) + ((u>>1)&0x55555555)); - u = ((u&0x33333333) + ((u>>2)&0x33333333)); - u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f)); - u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff)); - u = ( u&0x0000ffff) + (u>>16); - return u; -} - -static inline uint32_t lsbindex (uint32_t u) -{ - return popcount ((u&-u)-1); -} - -#define IOCTL(args) do { \ - int ret = ioctl args; \ - if (-1 == ret) { \ - ERRFail (#args); \ - } \ - ldebug ("ioctl " #args " = %d\n", ret); \ -} while (0) - -typedef struct AudioState { - int fd; - int freq; - int bits16; - int nchannels; - int rpos; - int wpos; - int live; - int oss_fmt; - int bytes_per_second; - int is_mapped; - void *buf; - int bufsize; - int nfrags; - int fragsize; - int old_optr; - int leftover; - uint64_t old_ticks; - void (*copy_fn)(void *, void *, int); -} AudioState; - -static AudioState oss_audio = { .fd = -1 }; - -static struct { - int try_mmap; - int nfrags; - int fragsize; -} conf = { - .try_mmap = 0, - .nfrags = 4, - .fragsize = 4096 -}; - -static enum {DONT, DSP, TID} est = DONT; - -static void pab (AudioState *s, struct audio_buf_info *abinfo) -{ - DEREF (abinfo); - - ldebug ("fragments %d, fragstotal %d, fragsize %d, bytes %d\n" - "rpos %d, wpos %d, live %d\n", - abinfo->fragments, - abinfo->fragstotal, - abinfo->fragsize, - abinfo->bytes, - s->rpos, s->wpos, s->live); -} - -static void do_open (AudioState *s) -{ - int mmmmssss; - audio_buf_info abinfo; - int fmt, freq, nchannels; - - if (s->buf) { - if (s->is_mapped) { - if (-1 == munmap (s->buf, s->bufsize)) { - ERRFail ("failed to unmap audio buffer %p %d", - s->buf, s->bufsize); - } - } - else { - qemu_free (s->buf); - } - s->buf = NULL; - } - - if (-1 != s->fd) - close (s->fd); - - s->fd = open ("/dev/dsp", O_RDWR | O_NONBLOCK); - if (-1 == s->fd) { - ERRFail ("can not open /dev/dsp"); - } - - fmt = s->oss_fmt; - freq = s->freq; - nchannels = s->nchannels; - - IOCTL ((s->fd, SNDCTL_DSP_RESET, 1)); - IOCTL ((s->fd, SNDCTL_DSP_SAMPLESIZE, &fmt)); - IOCTL ((s->fd, SNDCTL_DSP_CHANNELS, &nchannels)); - IOCTL ((s->fd, SNDCTL_DSP_SPEED, &freq)); - IOCTL ((s->fd, SNDCTL_DSP_NONBLOCK)); - - mmmmssss = (conf.nfrags << 16) | conf.fragsize; - IOCTL ((s->fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)); - - if ((s->oss_fmt != fmt) - || (s->nchannels != nchannels) - || (s->freq != freq)) { - Fail ("failed to set audio parameters\n" - "parameter | requested value | obtained value\n" - "format | %10d | %10d\n" - "channels | %10d | %10d\n" - "frequency | %10d | %10d\n", - s->oss_fmt, fmt, - s->nchannels, nchannels, - s->freq, freq); - } - - IOCTL ((s->fd, SNDCTL_DSP_GETOSPACE, &abinfo)); - - s->nfrags = abinfo.fragstotal; - s->fragsize = abinfo.fragsize; - s->bufsize = s->nfrags * s->fragsize; - s->old_optr = 0; - - s->bytes_per_second = (freq << (nchannels >> 1)) << s->bits16; - - linfo ("bytes per second %d\n", s->bytes_per_second); - - linfo ("fragments %d, fragstotal %d, fragsize %d, bytes %d, bufsize %d\n", - abinfo.fragments, - abinfo.fragstotal, - abinfo.fragsize, - abinfo.bytes, - s->bufsize); - - s->buf = MAP_FAILED; - s->is_mapped = 0; - - if (conf.try_mmap) { - s->buf = mmap (NULL, s->bufsize, PROT_WRITE, MAP_SHARED, s->fd, 0); - if (MAP_FAILED == s->buf) { - int err; - - err = errno; - dolog ("failed to mmap audio, size %d, fd %d\n" - "syserr: %s\n", - s->bufsize, s->fd, strerror (err)); - } - else { - est = TID; - s->is_mapped = 1; - } - } - - if (MAP_FAILED == s->buf) { - est = TID; - s->buf = qemu_mallocz (s->bufsize); - if (!s->buf) { - ERRFail ("audio buf malloc failed, size %d", s->bufsize); - } - } - - s->rpos = 0; - s->wpos = 0; - s->live = 0; - - if (s->is_mapped) { - int trig; - - trig = 0; - IOCTL ((s->fd, SNDCTL_DSP_SETTRIGGER, &trig)); - trig = PCM_ENABLE_OUTPUT; - IOCTL ((s->fd, SNDCTL_DSP_SETTRIGGER, &trig)); - } -} - -static void maybe_open (AudioState *s, int req_freq, int req_nchannels, - audfmt_e req_fmt, int force_open) -{ - int oss_fmt, bits16; - - switch (req_fmt) { - case AUD_FMT_U8: - bits16 = 0; - oss_fmt = AFMT_U8; - s->copy_fn = copy_no_conversion; - break; - - case AUD_FMT_S8: - Fail ("can not play 8bit signed"); - - case AUD_FMT_S16: - bits16 = 1; - oss_fmt = AFMT_S16_LE; - s->copy_fn = copy_no_conversion; - break; - - case AUD_FMT_U16: - bits16 = 1; - oss_fmt = AFMT_S16_LE; - s->copy_fn = copy_u16_to_s16; - break; - - default: - abort (); - } - - if (force_open - || (-1 == s->fd) - || (oss_fmt != s->oss_fmt) - || (req_nchannels != s->nchannels) - || (req_freq != s->freq) - || (bits16 != s->bits16)) { - s->oss_fmt = oss_fmt; - s->nchannels = req_nchannels; - s->freq = req_freq; - s->bits16 = bits16; - do_open (s); - } -} - -void AUD_reset (int req_freq, int req_nchannels, audfmt_e req_fmt) -{ - AudioState *s = &oss_audio; - maybe_open (s, req_freq, req_nchannels, req_fmt, 0); -} - -void AUD_open (int req_freq, int req_nchannels, audfmt_e req_fmt) -{ - AudioState *s = &oss_audio; - maybe_open (s, req_freq, req_nchannels, req_fmt, 1); -} - -int AUD_write (void *in_buf, int size) -{ - AudioState *s = &oss_audio; - int to_copy, temp; - uint8_t *in, *out; - - to_copy = MIN (s->bufsize - s->live, size); - - temp = to_copy; - - in = in_buf; - out = s->buf; - - while (temp) { - int copy; - - copy = MIN (temp, s->bufsize - s->wpos); - s->copy_fn (out + s->wpos, in, copy); - - s->wpos += copy; - if (s->wpos == s->bufsize) { - s->wpos = 0; - } - - temp -= copy; - in += copy; - s->live += copy; - } - - return to_copy; -} - -void AUD_run (void) -{ - int res; - int bytes; - struct audio_buf_info abinfo; - AudioState *s = &oss_audio; - - if (0 == s->live) - return; - - if (s->is_mapped) { - count_info info; - - res = ioctl (s->fd, SNDCTL_DSP_GETOPTR, &info); - if (res < 0) { - int err; - - err = errno; - lwarn ("SNDCTL_DSP_GETOPTR failed with %s\n", strerror (err)); - return; - } - - if (info.ptr > s->old_optr) { - bytes = info.ptr - s->old_optr; - } - else { - bytes = s->bufsize + info.ptr - s->old_optr; - } - - s->old_optr = info.ptr; - s->live -= bytes; - return; - } - - res = ioctl (s->fd, SNDCTL_DSP_GETOSPACE, &abinfo); - - if (res < 0) { - int err; - - err = errno; - lwarn ("SNDCTL_DSP_GETOSPACE failed with %s\n", strerror (err)); - return; - } - - bytes = abinfo.bytes; - bytes = MIN (s->live, bytes); -#if 0 - bytes = (bytes / fragsize) * fragsize; -#endif - - while (bytes) { - int left, play, written; - - left = s->bufsize - s->rpos; - play = MIN (left, bytes); - written = write (s->fd, (uint8_t *)s->buf + s->rpos, play); - - if (-1 == written) { - if (EAGAIN == errno || EINTR == errno) { - return; - } - else { - ERRFail ("write audio"); - } - } - - play = written; - s->live -= play; - s->rpos += play; - bytes -= play; - - if (s->rpos == s->bufsize) { - s->rpos = 0; - } - } -} - -static int get_dsp_bytes (void) -{ - int res; - struct count_info info; - AudioState *s = &oss_audio; - - res = ioctl (s->fd, SNDCTL_DSP_GETOPTR, &info); - if (-1 == res) { - int err; - - err = errno; - lwarn ("SNDCTL_DSP_GETOPTR failed with %s\n", strerror (err)); - return -1; - } - else { - ldebug ("bytes %d\n", info.bytes); - return info.bytes; - } -} - -void AUD_adjust_estimate (int leftover) -{ - AudioState *s = &oss_audio; - s->leftover = leftover; -} - -int AUD_get_free (void) -{ - int free, elapsed; - AudioState *s = &oss_audio; - - free = s->bufsize - s->live; - - if (free <= 0) - return 0; - - elapsed = free; - switch (est) { - case DONT: - break; - - case DSP: - { - static int old_bytes; - int bytes; - - bytes = get_dsp_bytes (); - if (bytes <= 0) - return free; - - elapsed = bytes - old_bytes; - old_bytes = bytes; - ldebug ("dsp elapsed %d bytes\n", elapsed); - break; - } - - case TID: - { - uint64_t ticks, delta; - uint64_t ua_elapsed; - uint64_t al_elapsed; - - ticks = qemu_get_clock(rt_clock); - delta = ticks - s->old_ticks; - s->old_ticks = ticks; - - ua_elapsed = (delta * s->bytes_per_second) / 1000; - al_elapsed = ua_elapsed & ~3ULL; - - ldebug ("tid elapsed %llu bytes\n", ua_elapsed); - - if (al_elapsed > (uint64_t) INT_MAX) - elapsed = INT_MAX; - else - elapsed = al_elapsed; - - elapsed += s->leftover; - } - } - - if (elapsed > free) { - lwarn ("audio can not keep up elapsed %d free %d\n", elapsed, free); - return free; - } - else { - return elapsed; - } -} - -int AUD_get_live (void) -{ - AudioState *s = &oss_audio; - return s->live; -} - -int AUD_get_buffer_size (void) -{ - AudioState *s = &oss_audio; - return s->bufsize; -} - -#define QC_OSS_FRAGSIZE "QEMU_OSS_FRAGSIZE" -#define QC_OSS_NFRAGS "QEMU_OSS_NFRAGS" -#define QC_OSS_MMAP "QEMU_OSS_MMAP" - -void AUD_init (void) -{ - int fsp; - - DEREF (pab); - - conf.fragsize = get_conf_val (QC_OSS_FRAGSIZE, conf.fragsize); - conf.nfrags = get_conf_val (QC_OSS_NFRAGS, conf.nfrags); - conf.try_mmap = get_conf_val (QC_OSS_MMAP, conf.try_mmap); - - fsp = conf.fragsize; - if (0 != (fsp & (fsp - 1))) { - Fail ("fragment size %d is not power of 2", fsp); - } - - conf.fragsize = lsbindex (fsp); -} - -#else - -void AUD_run (void) -{ -} - -int AUD_write (void *in_buf, int size) -{ - return 0; -} - -void AUD_reset (int rfreq, int rnchannels, audfmt_e rfmt) -{ -} - -void AUD_adjust_estimate (int _leftover) -{ -} - -int AUD_get_free (void) -{ - return 0; -} - -int AUD_get_live (void) -{ - return 0; -} - -int AUD_get_buffer_size (void) -{ - return 0; -} - -void AUD_init (void) -{ -} - -#endif @@ -2438,12 +2438,6 @@ void main_loop_wait(int timeout) if (vm_running) { qemu_run_timers(&active_timers[QEMU_TIMER_VIRTUAL], qemu_get_clock(vm_clock)); - - if (audio_enabled) { - /* XXX: add explicit timer */ - SB16_run(); - } - /* run dma transfers, if any */ DMA_run(); } @@ -30,6 +30,7 @@ #include <stdarg.h> #include <string.h> #include <inttypes.h> +#include <limits.h> #include <time.h> #include <ctype.h> #include <errno.h> @@ -553,39 +554,22 @@ void pci_piix3_ide_init(PCIBus *bus, BlockDriverState **hd_table); int pmac_ide_init (BlockDriverState **hd_table, openpic_t *openpic, int irq); -/* oss.c */ -typedef enum { - AUD_FMT_U8, - AUD_FMT_S8, - AUD_FMT_U16, - AUD_FMT_S16 -} audfmt_e; - -void AUD_open (int rfreq, int rnchannels, audfmt_e rfmt); -void AUD_reset (int rfreq, int rnchannels, audfmt_e rfmt); -int AUD_write (void *in_buf, int size); -void AUD_run (void); -void AUD_adjust_estimate (int _leftover); -int AUD_get_free (void); -int AUD_get_live (void); -int AUD_get_buffer_size (void); +/* audio.c */ void AUD_init (void); /* dma.c */ -typedef int (*DMA_transfer_handler) (void *opaque, target_ulong addr, int size); +typedef int (*DMA_transfer_handler) (void *opaque, int nchan, int pos, int size); int DMA_get_channel_mode (int nchan); +int DMA_read_memory (int nchan, void *buf, int pos, int size); +int DMA_write_memory (int nchan, void *buf, int pos, int size); void DMA_hold_DREQ (int nchan); void DMA_release_DREQ (int nchan); void DMA_schedule(int nchan); void DMA_run (void); void DMA_init (int high_page_enable); void DMA_register_channel (int nchan, - DMA_transfer_handler transfer_handler, void *opaque); - -/* sb16.c */ -void SB16_run (void); -void SB16_init (void); - + DMA_transfer_handler transfer_handler, + void *opaque); /* fdc.c */ #define MAX_FD 2 extern BlockDriverState *fd_table[MAX_FD]; |