Archipelago » qemu

/*

 * 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 "vl.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

};