| /* |
| * 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 |
| #endif |
| #include <signal.h> |
| #endif |
| |
| #define AUDIO_CAP "sdl" |
| #include "audio_int.h" |
| |
| typedef struct SDLVoiceOut { |
| HWVoiceOut hw; |
| int live; |
| int rpos; |
| int decr; |
| } SDLVoiceOut; |
| |
| 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 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; |
| } |
| |
| 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); |
| } |
| |
| 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; |
| sdl_unlock_and_post (s, "sdl_close"); |
| SDL_PauseAudio (1); |
| SDL_CloseAudio (); |
| s->initialized = 0; |
| } |
| } |
| |
| 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); */ |
| } |
| |
| static int sdl_write_out (SWVoiceOut *sw, void *buf, int len) |
| { |
| return audio_pcm_sw_write (sw, buf, len); |
| } |
| |
| 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; |
| } |
| |
| static void sdl_fini_out (HWVoiceOut *hw) |
| { |
| (void) hw; |
| |
| sdl_close (&glob_sdl); |
| } |
| |
| 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 (sdl_open (&req, &obt)) { |
| return -1; |
| } |
| |
| 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; |
| |
| 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; |
| } |
| |
| 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; |
| } |
| |
| 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); |
| } |
| |
| 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 http://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 |
| }; |