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-rw-r--r--audio/sdlaudio.c660
1 files changed, 660 insertions, 0 deletions
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
new file mode 100644
index 0000000..ea5bccd
--- /dev/null
+++ b/audio/sdlaudio.c
@@ -0,0 +1,660 @@
+/*
+ * 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
+};
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