/* * 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> #include "qemu-common.h" #include "audio.h" #ifndef _WIN32 #ifdef __sun__ #define _POSIX_PTHREAD_SEMANTICS 1 #elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__) #include <pthread.h> #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; static 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); struct st_sample *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; struct st_sample* 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, struct audsettings *as) { SDLVoiceOut *sdl = (SDLVoiceOut *) hw; SDLAudioState *s = &glob_sdl; SDL_AudioSpec req, obt; int shift; int endianess; int err; audfmt_e effective_fmt; struct audsettings 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 };