diff options
Diffstat (limited to 'audio')
| -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 |
13 files changed, 3125 insertions, 0 deletions
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 */ |