/*
* QEMU SDL audio driver
*
* Copyright (c) 2004-2005 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.h>
#include <SDL_thread.h>
#ifndef _WIN32
#ifdef __sun__
#define _POSIX_PTHREAD_SEMANTICS 1
#endif
#include <signal.h>
#endif
#define AUDIO_CAP "sdl"
#include "audio_int.h"
/* define DEBUG to 1 to dump audio debugging info at runtime to stderr */
#define DEBUG 0
/* define NEW_AUDIO to 1 to activate the new audio thread callback */
#define NEW_AUDIO 1
#if DEBUG
# define D(...) fprintf(stderr, __VA_ARGS__)
#else
# define D(...) ((void)0)
#endif
static struct {
int nb_samples;
} conf = {
1024
};
#if DEBUG
int64_t start_time;
#endif
#if NEW_AUDIO
#define AUDIO_BUFFER_SIZE (8192)
typedef HWVoiceOut SDLVoiceOut;
struct SDLAudioState {
int exit;
SDL_mutex* mutex;
int initialized;
uint8_t data[ AUDIO_BUFFER_SIZE ];
int pos, count;
} glob_sdl;
#else /* !NEW_AUDIO */
typedef struct SDLVoiceOut {
HWVoiceOut hw;
int live;
int rpos;
int decr;
} SDLVoiceOut;
struct SDLAudioState {
int exit;
SDL_mutex *mutex;
SDL_sem *sem;
int initialized;
} glob_sdl;
#endif /* !NEW_AUDIO */
typedef struct SDLAudioState SDLAudioState;
static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
}
static int sdl_lock (SDLAudioState *s, const char *forfn)
{
if (SDL_LockMutex (s->mutex)) {
sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
return -1;
}
return 0;
}
static int sdl_unlock (SDLAudioState *s, const char *forfn)
{
if (SDL_UnlockMutex (s->mutex)) {
sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
return -1;
}
return 0;
}
#if !NEW_AUDIO
static int sdl_post (SDLAudioState *s, const char *forfn)
{
if (SDL_SemPost (s->sem)) {
sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
return -1;
}
return 0;
}
static int sdl_wait (SDLAudioState *s, const char *forfn)
{
if (SDL_SemWait (s->sem)) {
sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
return -1;
}
return 0;
}
static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
{
if (sdl_unlock (s, forfn)) {
return -1;
}
return sdl_post (s, forfn);
}
#endif
static int aud_to_sdlfmt (audfmt_e fmt, int *shift)
{
switch (fmt) {
case AUD_FMT_S8:
*shift = 0;
return AUDIO_S8;
case AUD_FMT_U8:
*shift = 0;
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\n", fmt);
#ifdef DEBUG_AUDIO
abort ();
#endif
return AUDIO_U8;
}
}
static int sdl_to_audfmt (int sdlfmt, audfmt_e *fmt, int *endianess)
{
switch (sdlfmt) {
case AUDIO_S8:
*endianess = 0;
*fmt = AUD_FMT_S8;
break;
case AUDIO_U8:
*endianess = 0;
*fmt = AUD_FMT_U8;
break;
case AUDIO_S16LSB:
*endianess = 0;
*fmt = AUD_FMT_S16;
break;
case AUDIO_U16LSB:
*endianess = 0;
*fmt = AUD_FMT_U16;
break;
case AUDIO_S16MSB:
*endianess = 1;
*fmt = AUD_FMT_S16;
break;
case AUDIO_U16MSB:
*endianess = 1;
*fmt = AUD_FMT_U16;
break;
default:
dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
return -1;
}
return 0;
}
static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
{
int status;
#ifndef _WIN32
sigset_t new, old;
/* Make sure potential threads created by SDL don't hog signals. */
sigfillset (&new);
pthread_sigmask (SIG_BLOCK, &new, &old);
#endif
status = SDL_OpenAudio (req, obt);
if (status) {
sdl_logerr ("SDL_OpenAudio failed\n");
}
#ifndef _WIN32
pthread_sigmask (SIG_SETMASK, &old, 0);
#endif
return status;
}
static void sdl_close (SDLAudioState *s)
{
if (s->initialized) {
sdl_lock (s, "sdl_close");
s->exit = 1;
#if NEW_AUDIO
sdl_unlock (s, "sdl_close");
#else
sdl_unlock_and_post (s, "sdl_close");
#endif
SDL_PauseAudio (1);
SDL_CloseAudio ();
s->initialized = 0;
}
}
#if NEW_AUDIO
static void sdl_callback (void *opaque, Uint8 *buf, int len)
{
#if DEBUG
int64_t now;
#endif
SDLAudioState *s = &glob_sdl;
if (s->exit) {
return;
}
sdl_lock (s, "sdl_callback");
#if DEBUG
if (s->count > 0) {
now = qemu_get_clock(vm_clock);
if (start_time == 0)
start_time = now;
now = now - start_time;
D( "R %6.3f: pos:%5d count:%5d len:%5d\n", now/1e9, s->pos, s->count, len );
}
#endif
while (len > 0) {
int avail = audio_MIN( AUDIO_BUFFER_SIZE - s->pos, s->count );
if (avail == 0)
break;
if (avail > len)
avail = len;
memcpy( buf, s->data + s->pos, avail );
buf += avail;
len -= avail;
s->count -= avail;
s->pos += avail;
if (s->pos == AUDIO_BUFFER_SIZE)
s->pos = 0;
}
sdl_unlock (s, "sdl_callback");
}
#else /* !NEW_AUDIO */
static void sdl_callback (void *opaque, Uint8 *buf, int len)
{
SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
HWVoiceOut *hw = &sdl->hw;
int samples = len >> hw->info.shift;
if (s->exit) {
return;
}
while (samples) {
int to_mix, decr;
/* dolog ("in callback samples=%d\n", samples); */
sdl_wait (s, "sdl_callback");
if (s->exit) {
return;
}
if (sdl_lock (s, "sdl_callback")) {
return;
}
if (audio_bug (AUDIO_FUNC, sdl->live < 0 || sdl->live > hw->samples)) {
dolog ("sdl->live=%d hw->samples=%d\n",
sdl->live, hw->samples);
return;
}
if (!sdl->live) {
goto again;
}
/* dolog ("in callback live=%d\n", live); */
to_mix = audio_MIN (samples, sdl->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);
sdl->rpos = (sdl->rpos + chunk) % hw->samples;
to_mix -= chunk;
buf += chunk << hw->info.shift;
}
samples -= decr;
sdl->live -= decr;
sdl->decr += decr;
again:
if (sdl_unlock (s, "sdl_callback")) {
return;
}
}
/* dolog ("done len=%d\n", len); */
}
#endif /* !NEW_AUDIO */
static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}
#if NEW_AUDIO
static int sdl_run_out (HWVoiceOut *hw)
{
SDLAudioState *s = &glob_sdl;
int live, avail, end, total;
if (sdl_lock (s, "sdl_run_out")) {
return 0;
}
avail = AUDIO_BUFFER_SIZE - s->count;
end = s->pos + s->count;
if (end >= AUDIO_BUFFER_SIZE)
end -= AUDIO_BUFFER_SIZE;
sdl_unlock (s, "sdl_run_out");
live = audio_pcm_hw_get_live_out (hw);
total = 0;
while (live > 0) {
int bytes = audio_MIN(AUDIO_BUFFER_SIZE - end, avail);
int samples = bytes >> hw->info.shift;
int hwsamples = audio_MIN(hw->samples - hw->rpos, live);
uint8_t* dst = s->data + end;
st_sample_t* src = hw->mix_buf + hw->rpos;
if (samples == 0)
break;
if (samples > hwsamples) {
samples = hwsamples;
bytes = hwsamples << hw->info.shift;
}
hw->clip (dst, src, samples);
hw->rpos += samples;
if (hw->rpos == hw->samples)
hw->rpos = 0;
live -= samples;
avail -= bytes;
end += bytes;
if (end == AUDIO_BUFFER_SIZE)
end = 0;
total += bytes;
}
sdl_lock (s, "sdl_run_out");
s->count += total;
sdl_unlock (s, "sdl_run_out");
return total >> hw->info.shift;
}
#else /* !NEW_AUDIO */
static int sdl_run_out (HWVoiceOut *hw)
{
int decr, live;
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDLAudioState *s = &glob_sdl;
if (sdl_lock (s, "sdl_callback")) {
return 0;
}
live = audio_pcm_hw_get_live_out (hw);
if (sdl->decr > live) {
ldebug ("sdl->decr %d live %d sdl->live %d\n",
sdl->decr,
live,
sdl->live);
}
decr = audio_MIN (sdl->decr, live);
sdl->decr -= decr;
sdl->live = live - decr;
hw->rpos = sdl->rpos;
if (sdl->live > 0) {
sdl_unlock_and_post (s, "sdl_callback");
}
else {
sdl_unlock (s, "sdl_callback");
}
return decr;
}
#endif /* !NEW_AUDIO */
static void sdl_fini_out (HWVoiceOut *hw)
{
(void) hw;
sdl_close (&glob_sdl);
}
#if DEBUG
typedef struct { int value; const char* name; } MatchRec;
typedef const MatchRec* Match;
static const char*
match_find( Match matches, int value, char* temp )
{
int nn;
for ( nn = 0; matches[nn].name != NULL; nn++ ) {
if ( matches[nn].value == value )
return matches[nn].name;
}
sprintf( temp, "(%d?)", value );
return temp;
}
static const MatchRec sdl_audio_format_matches[] = {
{ AUDIO_U8, "AUDIO_U8" },
{ AUDIO_S8, "AUDIO_S8" },
{ AUDIO_U16, "AUDIO_U16LE" },
{ AUDIO_S16, "AUDIO_S16LE" },
{ AUDIO_U16MSB, "AUDIO_U16BE" },
{ AUDIO_S16MSB, "AUDIO_S16BE" },
{ 0, NULL }
};
static void
print_sdl_audiospec( SDL_AudioSpec* spec, const char* prefix )
{
char temp[64];
const char* fmt;
if (!prefix)
prefix = "";
printf( "%s audiospec [freq:%d format:%s channels:%d samples:%d bytes:%d",
prefix,
spec->freq,
match_find( sdl_audio_format_matches, spec->format, temp ),
spec->channels,
spec->samples,
spec->size
);
printf( "]\n" );
}
#endif
static int sdl_init_out (HWVoiceOut *hw, audsettings_t *as)
{
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDLAudioState *s = &glob_sdl;
SDL_AudioSpec req, obt;
int shift;
int endianess;
int err;
audfmt_e effective_fmt;
audsettings_t obt_as;
shift <<= as->nchannels == 2;
req.freq = as->freq;
req.format = aud_to_sdlfmt (as->fmt, &shift);
req.channels = as->nchannels;
req.samples = conf.nb_samples;
req.callback = sdl_callback;
req.userdata = sdl;
#if DEBUG
print_sdl_audiospec( &req, "wanted" );
#endif
if (sdl_open (&req, &obt)) {
return -1;
}
#if DEBUG
print_sdl_audiospec( &req, "obtained" );
#endif
err = sdl_to_audfmt (obt.format, &effective_fmt, &endianess);
if (err) {
sdl_close (s);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.channels;
obt_as.fmt = effective_fmt;
obt_as.endianness = endianess;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
#if DEBUG
start_time = qemu_get_clock(vm_clock);
#endif
s->initialized = 1;
s->exit = 0;
SDL_PauseAudio (0);
return 0;
}
static int sdl_ctl_out (HWVoiceOut *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;
if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
sdl_logerr ("SDL failed to initialize audio subsystem\n");
return NULL;
}
s->mutex = SDL_CreateMutex ();
if (!s->mutex) {
sdl_logerr ("Failed to create SDL mutex\n");
SDL_QuitSubSystem (SDL_INIT_AUDIO);
return NULL;
}
#if !NEW_AUDIO
s->sem = SDL_CreateSemaphore (0);
if (!s->sem) {
sdl_logerr ("Failed to create SDL semaphore\n");
SDL_DestroyMutex (s->mutex);
SDL_QuitSubSystem (SDL_INIT_AUDIO);
return NULL;
}
#endif
return s;
}
static void sdl_audio_fini (void *opaque)
{
SDLAudioState *s = opaque;
sdl_close (s);
#if !NEW_AUDIO
if (s->sem) {
SDL_DestroySemaphore (s->sem);
s->sem = NULL;
}
#endif
if (s->mutex) {
SDL_DestroyMutex (s->mutex);
s->mutex = NULL;
}
SDL_QuitSubSystem (SDL_INIT_AUDIO);
}
static struct audio_option sdl_options[] = {
{"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
"Size of SDL buffer in samples", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
};
static struct audio_pcm_ops sdl_pcm_ops = {
sdl_init_out,
sdl_fini_out,
sdl_run_out,
sdl_write_out,
sdl_ctl_out,
NULL,
NULL,
NULL,
NULL,
NULL
};
struct audio_driver sdl_audio_driver = {
INIT_FIELD (name = ) "sdl",
INIT_FIELD (descr = ) "SDL audio (www.libsdl.org)",
INIT_FIELD (options = ) sdl_options,
INIT_FIELD (init = ) sdl_audio_init,
INIT_FIELD (fini = ) sdl_audio_fini,
INIT_FIELD (pcm_ops = ) &sdl_pcm_ops,
INIT_FIELD (can_be_default = ) 1,
INIT_FIELD (max_voices_out = ) 1,
INIT_FIELD (max_voices_in = ) 0,
INIT_FIELD (voice_size_out = ) sizeof (SDLVoiceOut),
INIT_FIELD (voice_size_in = ) 0
};