symbian-qemu-0.9.1-12/qemu-symbian-svp/audio/sdlaudio.c
author Gareth Stockwell <gareth.stockwell@accenture.com>
Mon, 06 Sep 2010 16:25:43 +0100
changeset 107 3bc1a978be44
parent 1 2fb8b9db1c86
permissions -rw-r--r--
Fix for Bug 3671 - QEMU GDB stub listens on IPv6-only port on Windows 7 The connection string used by the GDB stub does not specify which version of the Internet Protocol should be used by the port on which it listens. On host platforms with IPv6 support, such as Windows 7, this means that the stub listens on an IPv6-only port. Since the GDB client uses IPv4, this means that the client cannot connect to 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 "qemu-common.h"
#include "audio.h"

#ifndef _WIN32
#ifdef __sun__
#define _POSIX_PTHREAD_SEMANTICS 1
#elif defined(__OpenBSD__) || defined(__FreeBSD__)
#include <pthread.h>
#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
};

static 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);
            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); */
}

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, 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 (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
};