audio merge (malc)

git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1125 c046a42c-6fe2-441c-8c8c-71466251a162

13 files changed, 3125 insertions, 0 deletions

diff --git a/audio/audio.c b/audio/audio.c
new file mode 100644
index 0000000..f55e1a2
--- /dev/null
+++ b/audio/audio.c
@@ -0,0 +1,935 @@
+/*
+ * QEMU Audio subsystem
+ * 
+ * Copyright (c) 2003-2004 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 <assert.h>
+#include <limits.h>
+#include "vl.h"
+
+#define AUDIO_CAP "audio"
+#include "audio/audio.h"
+
+#define USE_SDL_AUDIO
+#define USE_WAV_AUDIO
+
+#if defined __linux__ || (defined _BSD && !defined __APPLE__)
+#define USE_OSS_AUDIO
+#endif
+
+#ifdef USE_OSS_AUDIO
+#include "audio/ossaudio.h"
+#endif
+
+#ifdef USE_SDL_AUDIO
+#include "audio/sdlaudio.h"
+#endif
+
+#ifdef USE_WAV_AUDIO
+#include "audio/wavaudio.h"
+#endif
+
+#ifdef USE_FMOD_AUDIO
+#include "audio/fmodaudio.h"
+#endif
+
+#define QC_AUDIO_DRV    "QEMU_AUDIO_DRV"
+#define QC_VOICES    "QEMU_VOICES"
+#define QC_FIXED_FORMAT "QEMU_FIXED_FORMAT"
+#define QC_FIXED_FREQ   "QEMU_FIXED_FREQ"
+
+extern void SB16_init (void);
+
+#ifdef USE_ADLIB
+extern void Adlib_init (void);
+#endif
+
+#ifdef USE_GUS
+extern void GUS_init (void);
+#endif
+
+static void (*hw_ctors[]) (void) = {
+    SB16_init,
+#ifdef USE_ADLIB
+    Adlib_init,
+#endif
+#ifdef USE_GUS
+    GUS_init,
+#endif
+    NULL
+};
+
+static HWVoice *hw_voice;
+
+AudioState audio_state = {
+    1,                          /* use fixed settings */
+    44100,                      /* fixed frequency */
+    2,                          /* fixed channels */
+    AUD_FMT_S16,                /* fixed format */
+    1,                          /* number of hw voices */
+    -1                          /* voice size */
+};
+
+/* http://www.df.lth.se/~john_e/gems/gem002d.html */
+/* http://www.multi-platforms.com/Tips/PopCount.htm */
+uint32_t popcount (uint32_t u)
+{
+    u = ((u&0x55555555) + ((u>>1)&0x55555555));
+    u = ((u&0x33333333) + ((u>>2)&0x33333333));
+    u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
+    u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
+    u = ( u&0x0000ffff) + (u>>16);
+    return u;
+}
+
+inline uint32_t lsbindex (uint32_t u)
+{
+    return popcount ((u&-u)-1);
+}
+
+int audio_get_conf_int (const char *key, int defval)
+{
+    int val = defval;
+    char *strval;
+
+    strval = getenv (key);
+    if (strval) {
+        val = atoi (strval);
+    }
+
+    return val;
+}
+
+const char *audio_get_conf_str (const char *key, const char *defval)
+{
+    const char *val = getenv (key);
+    if (!val)
+        return defval;
+    else
+        return val;
+}
+
+void audio_log (const char *fmt, ...)
+{
+    va_list ap;
+    va_start (ap, fmt);
+    vfprintf (stderr, fmt, ap);
+    va_end (ap);
+}
+
+/*
+ * Soft Voice
+ */
+void pcm_sw_free_resources (SWVoice *sw)
+{
+    if (sw->buf) qemu_free (sw->buf);
+    if (sw->rate) st_rate_stop (sw->rate);
+    sw->buf = NULL;
+    sw->rate = NULL;
+}
+
+int pcm_sw_alloc_resources (SWVoice *sw)
+{
+    sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t));
+    if (!sw->buf)
+        return -1;
+
+    sw->rate = st_rate_start (sw->freq, sw->hw->freq);
+    if (!sw->rate) {
+        qemu_free (sw->buf);
+        sw->buf = NULL;
+        return -1;
+    }
+    return 0;
+}
+
+void pcm_sw_fini (SWVoice *sw)
+{
+    pcm_sw_free_resources (sw);
+}
+
+int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
+                 int nchannels, audfmt_e fmt)
+{
+    int bits = 8, sign = 0;
+
+    switch (fmt) {
+    case AUD_FMT_S8:
+        sign = 1;
+    case AUD_FMT_U8:
+        break;
+
+    case AUD_FMT_S16:
+        sign = 1;
+    case AUD_FMT_U16:
+        bits = 16;
+        break;
+    }
+
+    sw->hw = hw;
+    sw->freq = freq;
+    sw->fmt = fmt;
+    sw->nchannels = nchannels;
+    sw->shift = (nchannels == 2) + (bits == 16);
+    sw->align = (1 << sw->shift) - 1;
+    sw->left = 0;
+    sw->pos = 0;
+    sw->wpos = 0;
+    sw->live = 0;
+    sw->ratio = (sw->hw->freq * ((int64_t) INT_MAX)) / sw->freq;
+    sw->bytes_per_second = sw->freq << sw->shift;
+    sw->conv = mixeng_conv[nchannels == 2][sign][bits == 16];
+
+    pcm_sw_free_resources (sw);
+    return pcm_sw_alloc_resources (sw);
+}
+
+/* Hard voice */
+void pcm_hw_free_resources (HWVoice *hw)
+{
+    if (hw->mix_buf)
+        qemu_free (hw->mix_buf);
+    hw->mix_buf = NULL;
+}
+
+int pcm_hw_alloc_resources (HWVoice *hw)
+{
+    hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
+    if (!hw->mix_buf)
+        return -1;
+    return 0;
+}
+
+
+void pcm_hw_fini (HWVoice *hw)
+{
+    if (hw->active) {
+        ldebug ("pcm_hw_fini: %d %d %d\n", hw->freq, hw->nchannels, hw->fmt);
+        pcm_hw_free_resources (hw);
+        hw->pcm_ops->fini (hw);
+        memset (hw, 0, audio_state.drv->voice_size);
+    }
+}
+
+void pcm_hw_gc (HWVoice *hw)
+{
+    if (hw->nb_voices)
+        return;
+
+    pcm_hw_fini (hw);
+}
+
+int pcm_hw_get_live (HWVoice *hw)
+{
+    int i, alive = 0, live = hw->samples;
+
+    for (i = 0; i < hw->nb_voices; i++) {
+        if (hw->pvoice[i]->live) {
+            live = audio_MIN (hw->pvoice[i]->live, live);
+            alive += 1;
+        }
+    }
+
+    if (alive)
+        return live;
+    else
+        return -1;
+}
+
+int pcm_hw_get_live2 (HWVoice *hw, int *nb_active)
+{
+    int i, alive = 0, live = hw->samples;
+
+    *nb_active = 0;
+    for (i = 0; i < hw->nb_voices; i++) {
+        if (hw->pvoice[i]->live) {
+            if (hw->pvoice[i]->live < live) {
+                *nb_active = hw->pvoice[i]->active != 0;
+                live = hw->pvoice[i]->live;
+            }
+            alive += 1;
+        }
+    }
+
+    if (alive)
+        return live;
+    else
+        return -1;
+}
+
+void pcm_hw_dec_live (HWVoice *hw, int decr)
+{
+    int i;
+
+    for (i = 0; i < hw->nb_voices; i++) {
+        if (hw->pvoice[i]->live) {
+            hw->pvoice[i]->live -= decr;
+        }
+    }
+}
+
+void pcm_hw_clear (HWVoice *hw, void *buf, int len)
+{
+    if (!len)
+        return;
+
+    switch (hw->fmt) {
+    case AUD_FMT_S16:
+    case AUD_FMT_S8:
+        memset (buf, len << hw->shift, 0x00);
+        break;
+
+    case AUD_FMT_U8:
+        memset (buf, len << hw->shift, 0x80);
+        break;
+
+    case AUD_FMT_U16:
+        {
+            unsigned int i;
+            uint16_t *p = buf;
+            int shift = hw->nchannels - 1;
+
+            for (i = 0; i < len << shift; i++) {
+                p[i] = INT16_MAX;
+            }
+        }
+        break;
+    }
+}
+
+int pcm_hw_write (SWVoice *sw, void *buf, int size)
+{
+    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+    int ret = 0, pos = 0;
+    if (!sw)
+        return size;
+
+    hwsamples = sw->hw->samples;
+    samples = size >> sw->shift;
+
+    if (!sw->live) {
+        sw->wpos = sw->hw->rpos;
+    }
+    wpos = sw->wpos;
+    live = sw->live;
+    dead = hwsamples - live;
+    swlim = (dead * ((int64_t) INT_MAX)) / sw->ratio;
+    swlim = audio_MIN (swlim, samples);
+
+    ldebug ("size=%d live=%d dead=%d swlim=%d wpos=%d\n",
+           size, live, dead, swlim, wpos);
+    if (swlim)
+        sw->conv (sw->buf, buf, swlim);
+
+    while (swlim) {
+        dead = hwsamples - live;
+        left = hwsamples - wpos;
+        blck = audio_MIN (dead, left);
+        if (!blck) {
+            /* dolog ("swlim=%d\n", swlim); */
+            break;
+        }
+        isamp = swlim;
+        osamp = blck;
+        st_rate_flow (sw->rate, sw->buf + pos, sw->hw->mix_buf + wpos, &isamp, &osamp);
+        ret += isamp;
+        swlim -= isamp;
+        pos += isamp;
+        live += osamp;
+        wpos = (wpos + osamp) % hwsamples;
+    }
+
+    sw->wpos = wpos;
+    sw->live = live;
+    return ret << sw->shift;
+}
+
+int pcm_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    int sign = 0, bits = 8;
+
+    pcm_hw_fini (hw);
+    ldebug ("pcm_hw_init: %d %d %d\n", freq, nchannels, fmt);
+    if (hw->pcm_ops->init (hw, freq, nchannels, fmt)) {
+        memset (hw, 0, audio_state.drv->voice_size);
+        return -1;
+    }
+
+    switch (hw->fmt) {
+    case AUD_FMT_S8:
+        sign = 1;
+    case AUD_FMT_U8:
+        break;
+
+    case AUD_FMT_S16:
+        sign = 1;
+    case AUD_FMT_U16:
+        bits = 16;
+        break;
+    }
+
+    hw->nb_voices = 0;
+    hw->active = 1;
+    hw->shift = (hw->nchannels == 2) + (bits == 16);
+    hw->bytes_per_second = hw->freq << hw->shift;
+    hw->align = (1 << hw->shift) - 1;
+    hw->samples = hw->bufsize >> hw->shift;
+    hw->clip = mixeng_clip[hw->nchannels == 2][sign][bits == 16];
+    if (pcm_hw_alloc_resources (hw)) {
+        pcm_hw_fini (hw);
+        return -1;
+    }
+    return 0;
+}
+
+static int dist (void *hw)
+{
+    if (hw) {
+        return (((uint8_t *) hw - (uint8_t *) hw_voice)
+                / audio_state.voice_size) + 1;
+    }
+    else {
+        return 0;
+    }
+}
+
+#define ADVANCE(hw) hw ? advance (hw, audio_state.voice_size) : hw_voice
+
+HWVoice *pcm_hw_find_any (HWVoice *hw)
+{
+    int i = dist (hw);
+    for (; i < audio_state.nb_hw_voices; i++) {
+        hw = ADVANCE (hw);
+        return hw;
+    }
+    return NULL;
+}
+
+HWVoice *pcm_hw_find_any_active (HWVoice *hw)
+{
+    int i = dist (hw);
+    for (; i < audio_state.nb_hw_voices; i++) {
+        hw = ADVANCE (hw);
+        if (hw->active)
+            return hw;
+    }
+    return NULL;
+}
+
+HWVoice *pcm_hw_find_any_active_enabled (HWVoice *hw)
+{
+    int i = dist (hw);
+    for (; i < audio_state.nb_hw_voices; i++) {
+        hw = ADVANCE (hw);
+        if (hw->active && hw->enabled)
+            return hw;
+    }
+    return NULL;
+}
+
+HWVoice *pcm_hw_find_any_passive (HWVoice *hw)
+{
+    int i = dist (hw);
+    for (; i < audio_state.nb_hw_voices; i++) {
+        hw = ADVANCE (hw);
+        if (!hw->active)
+            return hw;
+    }
+    return NULL;
+}
+
+HWVoice *pcm_hw_find_specific (HWVoice *hw, int freq,
+                               int nchannels, audfmt_e fmt)
+{
+    while ((hw = pcm_hw_find_any_active (hw))) {
+        if (hw->freq == freq &&
+            hw->nchannels == nchannels &&
+            hw->fmt == fmt)
+            return hw;
+    }
+    return NULL;
+}
+
+HWVoice *pcm_hw_add (int freq, int nchannels, audfmt_e fmt)
+{
+    HWVoice *hw;
+
+    if (audio_state.fixed_format) {
+        freq = audio_state.fixed_freq;
+        nchannels = audio_state.fixed_channels;
+        fmt = audio_state.fixed_fmt;
+    }
+
+    hw = pcm_hw_find_specific (NULL, freq, nchannels, fmt);
+
+    if (hw)
+        return hw;
+
+    hw = pcm_hw_find_any_passive (NULL);
+    if (hw) {
+        hw->pcm_ops = audio_state.drv->pcm_ops;
+        if (!hw->pcm_ops)
+            return NULL;
+
+        if (pcm_hw_init (hw, freq, nchannels, fmt)) {
+            pcm_hw_gc (hw);
+            return NULL;
+        }
+        else
+            return hw;
+    }
+
+    return pcm_hw_find_any (NULL);
+}
+
+int pcm_hw_add_sw (HWVoice *hw, SWVoice *sw)
+{
+    SWVoice **pvoice = qemu_mallocz ((hw->nb_voices + 1) * sizeof (sw));
+    if (!pvoice)
+        return -1;
+
+    memcpy (pvoice, hw->pvoice, hw->nb_voices * sizeof (sw));
+    qemu_free (hw->pvoice);
+    hw->pvoice = pvoice;
+    hw->pvoice[hw->nb_voices++] = sw;
+    return 0;
+}
+
+int pcm_hw_del_sw (HWVoice *hw, SWVoice *sw)
+{
+    int i, j;
+    if (hw->nb_voices > 1) {
+        SWVoice **pvoice = qemu_mallocz ((hw->nb_voices - 1) * sizeof (sw));
+
+        if (!pvoice) {
+            dolog ("Can not maintain consistent state (not enough memory)\n");
+            return -1;
+        }
+
+        for (i = 0, j = 0; i < hw->nb_voices; i++) {
+            if (j >= hw->nb_voices - 1) {
+                dolog ("Can not maintain consistent state "
+                       "(invariant violated)\n");
+                return -1;
+            }
+            if (hw->pvoice[i] != sw)
+                pvoice[j++] = hw->pvoice[i];
+        }
+        qemu_free (hw->pvoice);
+        hw->pvoice = pvoice;
+        hw->nb_voices -= 1;
+    }
+    else {
+        qemu_free (hw->pvoice);
+        hw->pvoice = NULL;
+        hw->nb_voices = 0;
+    }
+    return 0;
+}
+
+SWVoice *pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt)
+{
+    SWVoice *sw;
+    HWVoice *hw;
+
+    sw = qemu_mallocz (sizeof (*sw));
+    if (!sw)
+        goto err1;
+
+    hw = pcm_hw_add (freq, nchannels, fmt);
+    if (!hw)
+        goto err2;
+
+    if (pcm_hw_add_sw (hw, sw))
+        goto err3;
+
+    if (pcm_sw_init (sw, hw, freq, nchannels, fmt))
+        goto err4;
+
+    return sw;
+
+err4:
+    pcm_hw_del_sw (hw, sw);
+err3:
+    pcm_hw_gc (hw);
+err2:
+    qemu_free (sw);
+err1:
+    return NULL;
+}
+
+SWVoice *AUD_open (SWVoice *sw, const char *name,
+                   int freq, int nchannels, audfmt_e fmt)
+{
+    if (!audio_state.drv) {
+        return NULL;
+    }
+
+    if (sw && freq == sw->freq && sw->nchannels == nchannels && sw->fmt == fmt) {
+        return sw;
+    }
+
+    if (sw) {
+        ldebug ("Different format %s %d %d %d\n",
+                name,
+                sw->freq == freq,
+                sw->nchannels == nchannels,
+                sw->fmt == fmt);
+    }
+
+    if (nchannels != 1 && nchannels != 2) {
+        dolog ("Bogus channel count %d for voice %s\n", nchannels, name);
+        return NULL;
+    }
+
+    if (!audio_state.fixed_format && sw) {
+        pcm_sw_fini (sw);
+        pcm_hw_del_sw (sw->hw, sw);
+        pcm_hw_gc (sw->hw);
+        if (sw->name) {
+            qemu_free (sw->name);
+            sw->name = NULL;
+        }
+        qemu_free (sw);
+        sw = NULL;
+    }
+
+    if (sw) {
+        HWVoice *hw = sw->hw;
+        if (!hw) {
+            dolog ("Internal logic error voice %s has no hardware store\n",
+                   name);
+            return sw;
+        }
+
+        if (pcm_sw_init (sw, hw, freq, nchannels, fmt)) {
+            pcm_sw_fini (sw);
+            pcm_hw_del_sw (hw, sw);
+            pcm_hw_gc (hw);
+            if (sw->name) {
+                qemu_free (sw->name);
+                sw->name = NULL;
+            }
+            qemu_free (sw);
+            return NULL;
+        }
+    }
+    else {
+        sw = pcm_create_voice_pair (freq, nchannels, fmt);
+        if (!sw) {
+            dolog ("Failed to create voice %s\n", name);
+            return NULL;
+        }
+    }
+
+    if (sw->name) {
+        qemu_free (sw->name);
+        sw->name = NULL;
+    }
+    sw->name = qemu_strdup (name);
+    return sw;
+}
+
+int AUD_write (SWVoice *sw, void *buf, int size)
+{
+    int bytes;
+
+    if (!sw->hw->enabled)
+        dolog ("Writing to disabled voice %s\n", sw->name);
+    bytes = sw->hw->pcm_ops->write (sw, buf, size);
+    return bytes;
+}
+
+void AUD_run (void)
+{
+    HWVoice *hw = NULL;
+
+    while ((hw = pcm_hw_find_any_active_enabled (hw))) {
+        int i;
+        if (hw->pending_disable && pcm_hw_get_live (hw) <= 0) {
+            hw->enabled = 0;
+            hw->pcm_ops->ctl (hw, VOICE_DISABLE);
+            for (i = 0; i < hw->nb_voices; i++) {
+                hw->pvoice[i]->live = 0;
+                /* hw->pvoice[i]->old_ticks = 0; */
+            }
+            continue;
+        }
+
+        hw->pcm_ops->run (hw);
+        assert (hw->rpos < hw->samples);
+        for (i = 0; i < hw->nb_voices; i++) {
+            SWVoice *sw = hw->pvoice[i];
+            if (!sw->active && !sw->live && sw->old_ticks) {
+                int64_t delta = qemu_get_clock (vm_clock) - sw->old_ticks;
+                if (delta > audio_state.ticks_threshold) {
+                    ldebug ("resetting old_ticks for %s\n", sw->name);
+                    sw->old_ticks = 0;
+                }
+            }
+        }
+    }
+}
+
+int AUD_get_free (SWVoice *sw)
+{
+    int free;
+
+    if (!sw)
+        return 4096;
+
+    free = ((sw->hw->samples - sw->live) << sw->hw->shift) * sw->ratio
+        / INT_MAX;
+
+    free &= ~sw->hw->align;
+    if (!free) return 0;
+
+    return free;
+}
+
+int AUD_get_buffer_size (SWVoice *sw)
+{
+    return sw->hw->bufsize;
+}
+
+void AUD_adjust (SWVoice *sw, int bytes)
+{
+    if (!sw)
+        return;
+    sw->old_ticks += (ticks_per_sec * (int64_t) bytes) / sw->bytes_per_second;
+}
+
+void AUD_reset (SWVoice *sw)
+{
+    sw->active = 0;
+    sw->old_ticks = 0;
+}
+
+int AUD_calc_elapsed (SWVoice *sw)
+{
+    int64_t now, delta, bytes;
+    int dead, swlim;
+
+    if (!sw)
+        return 0;
+
+    now = qemu_get_clock (vm_clock);
+    delta = now - sw->old_ticks;
+    bytes = (delta * sw->bytes_per_second) / ticks_per_sec;
+    if (delta < 0) {
+        dolog ("whoops delta(<0)=%lld\n", delta);
+        return 0;
+    }
+
+    dead = sw->hw->samples - sw->live;
+    swlim = ((dead * (int64_t) INT_MAX) / sw->ratio);
+
+    if (bytes > swlim) {
+        return swlim;
+    }
+    else {
+        return bytes;
+    }
+}
+
+void AUD_enable (SWVoice *sw, int on)
+{
+    int i;
+    HWVoice *hw;
+
+    if (!sw)
+        return;
+
+    hw = sw->hw;
+    if (on) {
+        if (!sw->live)
+            sw->wpos = sw->hw->rpos;
+        if (!sw->old_ticks) {
+            sw->old_ticks = qemu_get_clock (vm_clock);
+        }
+    }
+
+    if (sw->active != on) {
+        if (on) {
+            hw->pending_disable = 0;
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                for (i = 0; i < hw->nb_voices; i++) {
+                    ldebug ("resetting voice\n");
+                    sw = hw->pvoice[i];
+                    sw->old_ticks = qemu_get_clock (vm_clock);
+                }
+                hw->pcm_ops->ctl (hw, VOICE_ENABLE);
+            }
+        }
+        else {
+            if (hw->enabled && !hw->pending_disable) {
+                int nb_active = 0;
+                for (i = 0; i < hw->nb_voices; i++) {
+                    nb_active += hw->pvoice[i]->active != 0;
+                }
+
+                if (nb_active == 1) {
+                    hw->pending_disable = 1;
+                }
+            }
+        }
+        sw->active = on;
+    }
+}
+
+static struct audio_output_driver *drvtab[] = {
+#ifdef USE_OSS_AUDIO
+    &oss_output_driver,
+#endif
+#ifdef USE_FMOD_AUDIO
+    &fmod_output_driver,
+#endif
+#ifdef USE_SDL_AUDIO
+    &sdl_output_driver,
+#endif
+#ifdef USE_WAV_AUDIO
+    &wav_output_driver,
+#endif
+};
+
+static int voice_init (struct audio_output_driver *drv)
+{
+    audio_state.opaque = drv->init ();
+    if (audio_state.opaque) {
+        if (audio_state.nb_hw_voices > drv->max_voices) {
+            dolog ("`%s' does not support %d multiple hardware channels\n"
+                   "Resetting to %d\n",
+                   drv->name, audio_state.nb_hw_voices, drv->max_voices);
+            audio_state.nb_hw_voices = drv->max_voices;
+        }
+        hw_voice = qemu_mallocz (audio_state.nb_hw_voices * drv->voice_size);
+        if (hw_voice) {
+            audio_state.drv = drv;
+            return 1;
+        }
+        else {
+            dolog ("Not enough memory for %d `%s' voices (each %d bytes)\n",
+                   audio_state.nb_hw_voices, drv->name, drv->voice_size);
+            drv->fini (audio_state.opaque);
+            return 0;
+        }
+    }
+    else {
+        dolog ("Could not init `%s' audio\n", drv->name);
+        return 0;
+    }
+}
+
+static void audio_vm_stop_handler (void *opaque, int reason)
+{
+    HWVoice *hw = NULL;
+
+    while ((hw = pcm_hw_find_any (hw))) {
+        if (!hw->pcm_ops)
+            continue;
+
+        hw->pcm_ops->ctl (hw, reason ? VOICE_ENABLE : VOICE_DISABLE);
+    }
+}
+
+static void audio_atexit (void)
+{
+    HWVoice *hw = NULL;
+
+    while ((hw = pcm_hw_find_any (hw))) {
+        if (!hw->pcm_ops)
+            continue;
+
+        hw->pcm_ops->ctl (hw, VOICE_DISABLE);
+        hw->pcm_ops->fini (hw);
+    }
+    audio_state.drv->fini (audio_state.opaque);
+}
+
+static void audio_save (QEMUFile *f, void *opaque)
+{
+}
+
+static int audio_load (QEMUFile *f, void *opaque, int version_id)
+{
+    if (version_id != 1)
+        return -EINVAL;
+
+    return 0;
+}
+
+void AUD_init (void)
+{
+    int i;
+    int done = 0;
+    const char *drvname;
+
+    audio_state.fixed_format =
+        !!audio_get_conf_int (QC_FIXED_FORMAT, audio_state.fixed_format);
+    audio_state.fixed_freq =
+        audio_get_conf_int (QC_FIXED_FREQ, audio_state.fixed_freq);
+    audio_state.nb_hw_voices =
+        audio_get_conf_int (QC_VOICES, audio_state.nb_hw_voices);
+
+    if (audio_state.nb_hw_voices <= 0) {
+        dolog ("Bogus number of voices %d, resetting to 1\n",
+               audio_state.nb_hw_voices);
+    }
+
+    drvname = audio_get_conf_str (QC_AUDIO_DRV, NULL);
+    if (drvname) {
+        int found = 0;
+        for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+            if (!strcmp (drvname, drvtab[i]->name)) {
+                done = voice_init (drvtab[i]);
+                found = 1;
+                break;
+            }
+        }
+        if (!found) {
+            dolog ("Unknown audio driver `%s'\n", drvname);
+        }
+    }
+
+    qemu_add_vm_stop_handler (audio_vm_stop_handler, NULL);
+    atexit (audio_atexit);
+
+    if (!done) {
+        for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+            if (drvtab[i]->can_be_default)
+                done = voice_init (drvtab[i]);
+        }
+    }
+
+    audio_state.ticks_threshold = ticks_per_sec / 50;
+    audio_state.freq_threshold = 100;
+
+    register_savevm ("audio", 0, 1, audio_save, audio_load, NULL);
+    if (!done) {
+        dolog ("Can not initialize audio subsystem\n");
+        return;
+    }
+
+    for (i = 0; hw_ctors[i]; i++) {
+        hw_ctors[i] ();
+    }
+}
diff --git a/audio/audio.h b/audio/audio.h
new file mode 100644
index 0000000..926a1ba
--- /dev/null
+++ b/audio/audio.h
@@ -0,0 +1,188 @@
+/*
+ * QEMU Audio subsystem header
+ * 
+ * Copyright (c) 2003-2004 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.
+ */
+#ifndef QEMU_AUDIO_H
+#define QEMU_AUDIO_H
+
+#include "mixeng.h"
+
+#define dolog(...) fprintf (stderr, AUDIO_CAP ": " __VA_ARGS__)
+#ifdef DEBUG
+#define ldebug(...) dolog (__VA_ARGS__)
+#else
+#define ldebug(...)
+#endif
+
+typedef enum {
+  AUD_FMT_U8,
+  AUD_FMT_S8,
+  AUD_FMT_U16,
+  AUD_FMT_S16
+} audfmt_e;
+
+typedef struct HWVoice HWVoice;
+struct audio_output_driver;
+
+typedef struct AudioState {
+    int fixed_format;
+    int fixed_freq;
+    int fixed_channels;
+    int fixed_fmt;
+    int nb_hw_voices;
+    int voice_size;
+    int64_t ticks_threshold;
+    int freq_threshold;
+    void *opaque;
+    struct audio_output_driver *drv;
+} AudioState;
+
+extern AudioState audio_state;
+
+typedef struct SWVoice {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+
+    int shift;
+    int align;
+
+    t_sample *conv;
+
+    int left;
+    int pos;
+    int bytes_per_second;
+    int64_t ratio;
+    st_sample_t *buf;
+    void *rate;
+
+    int wpos;
+    int live;
+    int active;
+    int64_t old_ticks;
+    HWVoice *hw;
+    char *name;
+} SWVoice;
+
+#define VOICE_ENABLE 1
+#define VOICE_DISABLE 2
+
+struct pcm_ops {
+    int  (*init)  (HWVoice *hw, int freq, int nchannels, audfmt_e fmt);
+    void (*fini)  (HWVoice *hw);
+    void (*run)   (HWVoice *hw);
+    int  (*write) (SWVoice *sw, void *buf, int size);
+    int  (*ctl)   (HWVoice *hw, int cmd, ...);
+};
+
+struct audio_output_driver {
+    const char *name;
+    void *(*init) (void);
+    void (*fini) (void *);
+    struct pcm_ops *pcm_ops;
+    int can_be_default;
+    int max_voices;
+    int voice_size;
+};
+
+struct HWVoice {
+    int active;
+    int enabled;
+    int pending_disable;
+    int valid;
+    int freq;
+
+    f_sample *clip;
+    audfmt_e fmt;
+    int nchannels;
+
+    int align;
+    int shift;
+
+    int rpos;
+    int bufsize;
+
+    int bytes_per_second;
+    st_sample_t *mix_buf;
+
+    int samples;
+    int64_t old_ticks;
+    int nb_voices;
+    struct SWVoice **pvoice;
+    struct pcm_ops *pcm_ops;
+};
+
+void      audio_log (const char *fmt, ...);
+void      pcm_sw_free_resources (SWVoice *sw);
+int       pcm_sw_alloc_resources (SWVoice *sw);
+void      pcm_sw_fini (SWVoice *sw);
+int       pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
+                       int nchannels, audfmt_e fmt);
+
+void      pcm_hw_clear (HWVoice *hw, void *buf, int len);
+HWVoice * pcm_hw_find_any (HWVoice *hw);
+HWVoice * pcm_hw_find_any_active (HWVoice *hw);
+HWVoice * pcm_hw_find_any_passive (HWVoice *hw);
+HWVoice * pcm_hw_find_specific (HWVoice *hw, int freq,
+                                int nchannels, audfmt_e fmt);
+HWVoice * pcm_hw_add (int freq, int nchannels, audfmt_e fmt);
+int       pcm_hw_add_sw (HWVoice *hw, SWVoice *sw);
+int       pcm_hw_del_sw (HWVoice *hw, SWVoice *sw);
+SWVoice * pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt);
+
+void      pcm_hw_free_resources (HWVoice *hw);
+int       pcm_hw_alloc_resources (HWVoice *hw);
+void      pcm_hw_fini (HWVoice *hw);
+void      pcm_hw_gc (HWVoice *hw);
+int       pcm_hw_get_live (HWVoice *hw);
+int       pcm_hw_get_live2 (HWVoice *hw, int *nb_active);
+void      pcm_hw_dec_live (HWVoice *hw, int decr);
+int       pcm_hw_write (SWVoice *sw, void *buf, int len);
+
+int         audio_get_conf_int (const char *key, int defval);
+const char *audio_get_conf_str (const char *key, const char *defval);
+
+/* Public API */
+SWVoice * AUD_open (SWVoice *sw, const char *name, int freq,
+                    int nchannels, audfmt_e fmt);
+int    AUD_write (SWVoice *sw, void *pcm_buf, int size);
+void   AUD_adjust (SWVoice *sw, int leftover);
+void   AUD_reset (SWVoice *sw);
+int    AUD_get_free (SWVoice *sw);
+int    AUD_get_buffer_size (SWVoice *sw);
+void   AUD_run (void);
+void   AUD_enable (SWVoice *sw, int on);
+int    AUD_calc_elapsed (SWVoice *sw);
+
+static inline void *advance (void *p, int incr)
+{
+    uint8_t *d = p;
+    return (d + incr);
+}
+
+uint32_t popcount (uint32_t u);
+inline uint32_t lsbindex (uint32_t u);
+
+#define audio_MIN(a, b) ((a)>(b)?(b):(a))
+#define audio_MAX(a, b) ((a)<(b)?(b):(a))
+
+#endif  /* audio.h */
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
new file mode 100644
index 0000000..7457033
--- /dev/null
+++ b/audio/fmodaudio.c
@@ -0,0 +1,457 @@
+/*
+ * QEMU FMOD audio output driver
+ * 
+ * Copyright (c) 2004 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 <fmod.h>
+#include <fmod_errors.h>
+#include "vl.h"
+
+#define AUDIO_CAP "fmod"
+#include "audio/audio.h"
+#include "audio/fmodaudio.h"
+
+#define QC_FMOD_DRV "QEMU_FMOD_DRV"
+#define QC_FMOD_FREQ "QEMU_FMOD_FREQ"
+#define QC_FMOD_SAMPLES "QEMU_FMOD_SAMPLES"
+#define QC_FMOD_CHANNELS "QEMU_FMOD_CHANNELS"
+#define QC_FMOD_BUFSIZE "QEMU_FMOD_BUFSIZE"
+#define QC_FMOD_THRESHOLD "QEMU_FMOD_THRESHOLD"
+
+static struct {
+    int nb_samples;
+    int freq;
+    int nb_channels;
+    int bufsize;
+    int threshold;
+} conf = {
+    2048,
+    44100,
+    1,
+    0,
+    128
+};
+
+#define errstr() FMOD_ErrorString (FSOUND_GetError ())
+
+static int fmod_hw_write (SWVoice *sw, void *buf, int len)
+{
+    return pcm_hw_write (sw, buf, len);
+}
+
+static void fmod_clear_sample (FMODVoice *fmd)
+{
+    HWVoice *hw = &fmd->hw;
+    int status;
+    void *p1 = 0, *p2 = 0;
+    unsigned int len1 = 0, len2 = 0;
+
+    status = FSOUND_Sample_Lock (
+        fmd->fmod_sample,
+        0,
+        hw->samples << hw->shift,
+        &p1,
+        &p2,
+        &len1,
+        &len2
+        );
+
+    if (!status) {
+        dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+        return;
+    }
+
+    if ((len1 & hw->align) || (len2 & hw->align)) {
+        dolog ("Locking sample returned unaligned length %d, %d\n",
+               len1, len2);
+        goto fail;
+    }
+
+    if (len1 + len2 != hw->samples << hw->shift) {
+        dolog ("Locking sample returned incomplete length %d, %d\n",
+               len1 + len2, hw->samples << hw->shift);
+        goto fail;
+    }
+    pcm_hw_clear (hw, p1, hw->samples);
+
+ fail:
+    status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, len1, len2);
+    if (!status) {
+        dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+    }
+}
+
+static int fmod_write_sample (HWVoice *hw, uint8_t *dst, st_sample_t *src,
+                              int src_size, int src_pos, int dst_len)
+{
+    int src_len1 = dst_len, src_len2 = 0, pos = src_pos + dst_len;
+    st_sample_t *src1 = src + src_pos, *src2 = 0;
+
+    if (src_pos + dst_len > src_size) {
+        src_len1 = src_size - src_pos;
+        src2 = src;
+        src_len2 = dst_len - src_len1;
+        pos = src_len2;
+    }
+
+    if (src_len1) {
+        hw->clip (dst, src1, src_len1);
+        memset (src1, 0, src_len1 * sizeof (st_sample_t));
+        advance (dst, src_len1);
+    }
+
+    if (src_len2) {
+        hw->clip (dst, src2, src_len2);
+        memset (src2, 0, src_len2 * sizeof (st_sample_t));
+    }
+    return pos;
+}
+
+static int fmod_unlock_sample (FMODVoice *fmd, void *p1, void *p2,
+                               unsigned int blen1, unsigned int blen2)
+{
+    int status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, blen1, blen2);
+    if (!status) {
+        dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+        return -1;
+    }
+    return 0;
+}
+
+static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
+                             void **p1, void **p2,
+                             unsigned int *blen1, unsigned int *blen2)
+{
+    HWVoice *hw = &fmd->hw;
+    int status;
+
+    status = FSOUND_Sample_Lock (
+        fmd->fmod_sample,
+        pos << hw->shift,
+        len << hw->shift,
+        p1,
+        p2,
+        blen1,
+        blen2
+        );
+
+    if (!status) {
+        dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+        return -1;
+    }
+
+    if ((*blen1 & hw->align) || (*blen2 & hw->align)) {
+        dolog ("Locking sample returned unaligned length %d, %d\n",
+               *blen1, *blen2);
+        fmod_unlock_sample (fmd, *p1, *p2, *blen1, *blen2);
+        return -1;
+    }
+    return 0;
+}
+
+static void fmod_hw_run (HWVoice *hw)
+{
+    FMODVoice *fmd = (FMODVoice *) hw;
+    int rpos, live, decr;
+    void *p1 = 0, *p2 = 0;
+    unsigned int blen1 = 0, blen2 = 0;
+    unsigned int len1 = 0, len2 = 0;
+    int nb_active;
+
+    live = pcm_hw_get_live2 (hw, &nb_active);
+    if (live <= 0) {
+        return;
+    }
+
+    if (!hw->pending_disable
+        && nb_active
+        && conf.threshold
+        && live <= conf.threshold) {
+        ldebug ("live=%d nb_active=%d\n", live, nb_active);
+        return;
+    }
+
+    decr = live;
+
+#if 1
+    if (fmd->channel >= 0) {
+        int pos2 = (fmd->old_pos + decr) % hw->samples;
+        int pos = FSOUND_GetCurrentPosition (fmd->channel);
+
+        if (fmd->old_pos < pos && pos2 >= pos) {
+            decr = pos - fmd->old_pos - (pos2 == pos) - 1;
+        }
+        else if (fmd->old_pos > pos && pos2 >= pos && pos2 < fmd->old_pos) {
+            decr = (hw->samples - fmd->old_pos) + pos - (pos2 == pos) - 1;
+        }
+/*         ldebug ("pos=%d pos2=%d old=%d live=%d decr=%d\n", */
+/*                 pos, pos2, fmd->old_pos, live, decr); */
+    }
+#endif
+
+    if (decr <= 0) {
+        return;
+    }
+
+    if (fmod_lock_sample (fmd, fmd->old_pos, decr, &p1, &p2, &blen1, &blen2)) {
+        return;
+    }
+
+    len1 = blen1 >> hw->shift;
+    len2 = blen2 >> hw->shift;
+    ldebug ("%p %p %d %d %d %d\n", p1, p2, len1, len2, blen1, blen2);
+    decr = len1 + len2;
+    rpos = hw->rpos;
+
+    if (len1) {
+        rpos = fmod_write_sample (hw, p1, hw->mix_buf, hw->samples, rpos, len1);
+    }
+
+    if (len2) {
+        rpos = fmod_write_sample (hw, p2, hw->mix_buf, hw->samples, rpos, len2);
+    }
+
+    fmod_unlock_sample (fmd, p1, p2, blen1, blen2);
+
+    pcm_hw_dec_live (hw, decr);
+    hw->rpos = rpos % hw->samples;
+    fmd->old_pos = (fmd->old_pos + decr) % hw->samples;
+}
+
+static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
+{
+    int mode = FSOUND_LOOP_NORMAL;
+
+    switch (fmt) {
+    case AUD_FMT_S8:
+        mode |= FSOUND_SIGNED | FSOUND_8BITS;
+        break;
+
+    case AUD_FMT_U8:
+        mode |= FSOUND_UNSIGNED | FSOUND_8BITS;
+        break;
+
+    case AUD_FMT_S16:
+        mode |= FSOUND_SIGNED | FSOUND_16BITS;
+        break;
+
+    case AUD_FMT_U16:
+        mode |= FSOUND_UNSIGNED | FSOUND_16BITS;
+        break;
+
+    default:
+        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
+        exit (EXIT_FAILURE);
+    }
+    mode |= stereo ? FSOUND_STEREO : FSOUND_MONO;
+    return mode;
+}
+
+static void fmod_hw_fini (HWVoice *hw)
+{
+    FMODVoice *fmd = (FMODVoice *) hw;
+
+    if (fmd->fmod_sample) {
+        FSOUND_Sample_Free (fmd->fmod_sample);
+        fmd->fmod_sample = 0;
+
+        if (fmd->channel >= 0) {
+            FSOUND_StopSound (fmd->channel);
+        }
+    }
+}
+
+static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    int bits16, mode, channel;
+    FMODVoice *fmd = (FMODVoice *) hw;
+
+    mode = AUD_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+    fmd->fmod_sample = FSOUND_Sample_Alloc (
+        FSOUND_FREE,            /* index */
+        conf.nb_samples,        /* length */
+        mode,                   /* mode */
+        freq,                   /* freq */
+        255,                    /* volume */
+        128,                    /* pan */
+        255                     /* priority */
+        );
+
+    if (!fmd->fmod_sample) {
+        dolog ("Failed to allocate FMOD sample\nReason: %s\n", errstr ());
+        return -1;
+    }
+
+    channel = FSOUND_PlaySoundEx (FSOUND_FREE, fmd->fmod_sample, 0, 1);
+    if (channel < 0) {
+        dolog ("Failed to start playing sound\nReason: %s\n", errstr ());
+        FSOUND_Sample_Free (fmd->fmod_sample);
+        return -1;
+    }
+    fmd->channel = channel;
+
+    hw->freq = freq;
+    hw->fmt = fmt;
+    hw->nchannels = nchannels;
+    bits16 = fmt == AUD_FMT_U16 || fmt == AUD_FMT_S16;
+    hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
+    return 0;
+}
+
+static int fmod_hw_ctl (HWVoice *hw, int cmd, ...)
+{
+    int status;
+    FMODVoice *fmd = (FMODVoice *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        fmod_clear_sample (fmd);
+        status = FSOUND_SetPaused (fmd->channel, 0);
+        if (!status) {
+            dolog ("Failed to resume channel %d\nReason: %s\n",
+                   fmd->channel, errstr ());
+        }
+        break;
+
+    case VOICE_DISABLE:
+        status = FSOUND_SetPaused (fmd->channel, 1);
+        if (!status) {
+            dolog ("Failed to pause channel %d\nReason: %s\n",
+                   fmd->channel, errstr ());
+        }
+        break;
+    }
+    return 0;
+}
+
+static struct {
+    const char *name;
+    int type;
+} drvtab[] = {
+    {"none", FSOUND_OUTPUT_NOSOUND},
+#ifdef _WIN32
+    {"winmm", FSOUND_OUTPUT_WINMM},
+    {"dsound", FSOUND_OUTPUT_DSOUND},
+    {"a3d", FSOUND_OUTPUT_A3D},
+    {"asio", FSOUND_OUTPUT_ASIO},
+#endif
+#ifdef __linux__
+    {"oss", FSOUND_OUTPUT_OSS},
+    {"alsa", FSOUND_OUTPUT_ALSA},
+    {"esd", FSOUND_OUTPUT_ESD},
+#endif
+#ifdef __APPLE__
+    {"mac", FSOUND_OUTPUT_MAC},
+#endif
+#if 0
+    {"xbox", FSOUND_OUTPUT_XBOX},
+    {"ps2", FSOUND_OUTPUT_PS2},
+    {"gcube", FSOUND_OUTPUT_GC},
+#endif
+    {"nort", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
+};
+
+static void *fmod_audio_init (void)
+{
+    int i;
+    double ver;
+    int status;
+    int output_type = -1;
+    const char *drv = audio_get_conf_str (QC_FMOD_DRV, NULL);
+
+    ver = FSOUND_GetVersion ();
+    if (ver < FMOD_VERSION) {
+        dolog ("Wrong FMOD version %f, need at least %f\n", ver, FMOD_VERSION);
+        return NULL;
+    }
+
+    if (drv) {
+        int found = 0;
+        for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+            if (!strcmp (drv, drvtab[i].name)) {
+                output_type = drvtab[i].type;
+                found = 1;
+                break;
+            }
+        }
+        if (!found) {
+            dolog ("Unknown FMOD output driver `%s'\n", drv);
+        }
+    }
+
+    if (output_type != -1) {
+        status = FSOUND_SetOutput (output_type);
+        if (!status) {
+            dolog ("FSOUND_SetOutput(%d) failed\nReason: %s\n",
+                   output_type, errstr ());
+            return NULL;
+        }
+    }
+
+    conf.freq = audio_get_conf_int (QC_FMOD_FREQ, conf.freq);
+    conf.nb_samples = audio_get_conf_int (QC_FMOD_SAMPLES, conf.nb_samples);
+    conf.nb_channels =
+        audio_get_conf_int (QC_FMOD_CHANNELS,
+                            (audio_state.nb_hw_voices > 1
+                             ? audio_state.nb_hw_voices
+                             : conf.nb_channels));
+    conf.bufsize = audio_get_conf_int (QC_FMOD_BUFSIZE, conf.bufsize);
+    conf.threshold = audio_get_conf_int (QC_FMOD_THRESHOLD, conf.threshold);
+
+    if (conf.bufsize) {
+        status = FSOUND_SetBufferSize (conf.bufsize);
+        if (!status) {
+            dolog ("FSOUND_SetBufferSize (%d) failed\nReason: %s\n",
+                   conf.bufsize, errstr ());
+        }
+    }
+
+    status = FSOUND_Init (conf.freq, conf.nb_channels, 0);
+    if (!status) {
+        dolog ("FSOUND_Init failed\nReason: %s\n", errstr ());
+        return NULL;
+    }
+
+    return &conf;
+}
+
+static void fmod_audio_fini (void *opaque)
+{
+    FSOUND_Close ();
+}
+
+struct pcm_ops fmod_pcm_ops = {
+    fmod_hw_init,
+    fmod_hw_fini,
+    fmod_hw_run,
+    fmod_hw_write,
+    fmod_hw_ctl
+};
+
+struct audio_output_driver fmod_output_driver = {
+    "fmod",
+    fmod_audio_init,
+    fmod_audio_fini,
+    &fmod_pcm_ops,
+    1,
+    INT_MAX,
+    sizeof (FMODVoice)
+};
diff --git a/audio/fmodaudio.h b/audio/fmodaudio.h
new file mode 100644
index 0000000..9f85c30
--- /dev/null
+++ b/audio/fmodaudio.h
@@ -0,0 +1,39 @@
+/*
+ * QEMU FMOD audio output driver header
+ * 
+ * Copyright (c) 2004 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.
+ */
+#ifndef QEMU_FMODAUDIO_H
+#define QEMU_FMODAUDIO_H
+
+#include <fmod.h>
+
+typedef struct FMODVoice {
+    struct HWVoice hw;
+    unsigned int old_pos;
+    FSOUND_SAMPLE *fmod_sample;
+    int channel;
+} FMODVoice;
+
+extern struct pcm_ops fmod_pcm_ops;
+extern struct audio_output_driver fmod_output_driver;
+
+#endif  /* fmodaudio.h */
diff --git a/audio/mixeng.c b/audio/mixeng.c
new file mode 100644
index 0000000..b0bb412
--- /dev/null
+++ b/audio/mixeng.c
@@ -0,0 +1,255 @@
+/*
+ * QEMU Mixing engine
+ *
+ * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 1998 Fabrice Bellard
+ *
+ * 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 "vl.h"
+//#define DEBUG_FP
+#include "audio/mixeng.h"
+
+#define IN_T int8_t
+#define IN_MIN CHAR_MIN
+#define IN_MAX CHAR_MAX
+#define SIGNED
+#include "mixeng_template.h"
+#undef SIGNED
+#undef IN_MAX
+#undef IN_MIN
+#undef IN_T
+
+#define IN_T uint8_t
+#define IN_MIN 0
+#define IN_MAX UCHAR_MAX
+#include "mixeng_template.h"
+#undef IN_MAX
+#undef IN_MIN
+#undef IN_T
+
+#define IN_T int16_t
+#define IN_MIN SHRT_MIN
+#define IN_MAX SHRT_MAX
+#define SIGNED
+#include "mixeng_template.h"
+#undef SIGNED
+#undef IN_MAX
+#undef IN_MIN
+#undef IN_T
+
+#define IN_T uint16_t
+#define IN_MIN 0
+#define IN_MAX USHRT_MAX
+#include "mixeng_template.h"
+#undef IN_MAX
+#undef IN_MIN
+#undef IN_T
+
+t_sample *mixeng_conv[2][2][2] = {
+    {
+        {
+            conv_uint8_t_to_mono,
+            conv_uint16_t_to_mono
+        },
+        {
+            conv_int8_t_to_mono,
+            conv_int16_t_to_mono
+        }
+    },
+    {
+        {
+            conv_uint8_t_to_stereo,
+            conv_uint16_t_to_stereo
+        },
+        {
+            conv_int8_t_to_stereo,
+            conv_int16_t_to_stereo
+        }
+    }
+};
+
+f_sample *mixeng_clip[2][2][2] = {
+    {
+        {
+            clip_uint8_t_from_mono,
+            clip_uint16_t_from_mono
+        },
+        {
+            clip_int8_t_from_mono,
+            clip_int16_t_from_mono
+        }
+    },
+    {
+        {
+            clip_uint8_t_from_stereo,
+            clip_uint16_t_from_stereo
+        },
+        {
+            clip_int8_t_from_stereo,
+            clip_int16_t_from_stereo
+        }
+    }
+};
+
+/*
+ * August 21, 1998
+ * Copyright 1998 Fabrice Bellard.
+ *
+ * [Rewrote completly the code of Lance Norskog And Sundry
+ * Contributors with a more efficient algorithm.]
+ *
+ * This source code is freely redistributable and may be used for
+ * any purpose.  This copyright notice must be maintained. 
+ * Lance Norskog And Sundry Contributors are not responsible for 
+ * the consequences of using this software.  
+ */
+
+/*
+ * Sound Tools rate change effect file.
+ */
+/*
+ * Linear Interpolation.
+ *
+ * The use of fractional increment allows us to use no buffer. It
+ * avoid the problems at the end of the buffer we had with the old
+ * method which stored a possibly big buffer of size
+ * lcm(in_rate,out_rate).
+ *
+ * Limited to 16 bit samples and sampling frequency <= 65535 Hz. If
+ * the input & output frequencies are equal, a delay of one sample is
+ * introduced.  Limited to processing 32-bit count worth of samples.
+ *
+ * 1 << FRAC_BITS evaluating to zero in several places.  Changed with
+ * an (unsigned long) cast to make it safe.  MarkMLl 2/1/99
+ */
+
+/* Private data */
+typedef struct ratestuff {
+    uint64_t opos;
+    uint64_t opos_inc;
+    uint32_t ipos;              /* position in the input stream (integer) */
+    st_sample_t ilast;          /* last sample in the input stream */
+} *rate_t;
+
+/*
+ * Prepare processing.
+ */
+void *st_rate_start (int inrate, int outrate)
+{
+    rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
+
+    if (!rate) {
+        exit (EXIT_FAILURE);
+    }
+
+    if (inrate == outrate) {
+        // exit (EXIT_FAILURE);
+    }
+
+    if (inrate >= 65535 || outrate >= 65535) {
+        // exit (EXIT_FAILURE);
+    }
+
+    rate->opos = 0;
+
+    /* increment */
+    rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate;
+
+    rate->ipos = 0;
+    rate->ilast.l = 0;
+    rate->ilast.r = 0;
+    return rate;
+}
+
+/*
+ * Processed signed long samples from ibuf to obuf.
+ * Return number of samples processed.
+ */
+void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+                   int *isamp, int *osamp)
+{
+    rate_t rate = (rate_t) opaque;
+    st_sample_t *istart, *iend;
+    st_sample_t *ostart, *oend;
+    st_sample_t ilast, icur, out;
+    int64_t t;
+
+    ilast = rate->ilast;
+
+    istart = ibuf;
+    iend = ibuf + *isamp;
+
+    ostart = obuf;
+    oend = obuf + *osamp;
+
+    if (rate->opos_inc == 1ULL << 32) {
+        int i, n = *isamp > *osamp ? *osamp : *isamp;
+        for (i = 0; i < n; i++) {
+            obuf[i].l += ibuf[i].r;
+            obuf[i].r += ibuf[i].r;
+        }
+        *isamp = n;
+        *osamp = n;
+        return;
+    }
+
+    while (obuf < oend) {
+
+        /* Safety catch to make sure we have input samples.  */
+        if (ibuf >= iend)
+            break;
+
+        /* read as many input samples so that ipos > opos */
+
+        while (rate->ipos <= (rate->opos >> 32)) {
+            ilast = *ibuf++;
+            rate->ipos++;
+            /* See if we finished the input buffer yet */
+            if (ibuf >= iend) goto the_end;
+        }
+
+        icur = *ibuf;
+
+        /* interpolate */
+        t = rate->opos & 0xffffffff;
+        out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX;
+        out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX;
+
+        /* output sample & increment position */
+#if 0
+        *obuf++ = out;
+#else
+        obuf->l += out.l;
+        obuf->r += out.r;
+        obuf += 1;
+#endif
+        rate->opos += rate->opos_inc;
+    }
+
+the_end:
+    *isamp = ibuf - istart;
+    *osamp = obuf - ostart;
+    rate->ilast = ilast;
+}
+
+void st_rate_stop (void *opaque)
+{
+    qemu_free (opaque);
+}
diff --git a/audio/mixeng.h b/audio/mixeng.h
new file mode 100644
index 0000000..699435e
--- /dev/null
+++ b/audio/mixeng.h
@@ -0,0 +1,39 @@
+/*
+ * QEMU Mixing engine header
+ * 
+ * Copyright (c) 2004 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.
+ */
+#ifndef QEMU_MIXENG_H
+#define QEMU_MIXENG_H
+
+typedef void (t_sample) (void *dst, const void *src, int samples);
+typedef void (f_sample) (void *dst, const void *src, int samples);
+typedef struct { int64_t l; int64_t r; } st_sample_t;
+
+extern t_sample *mixeng_conv[2][2][2];
+extern f_sample *mixeng_clip[2][2][2];
+
+void *st_rate_start (int inrate, int outrate);
+void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+                   int *isamp, int *osamp);
+void st_rate_stop (void *opaque);
+
+#endif  /* mixeng.h */
diff --git a/audio/mixeng_template.h b/audio/mixeng_template.h
new file mode 100644
index 0000000..f3b3f65
--- /dev/null
+++ b/audio/mixeng_template.h
@@ -0,0 +1,111 @@
+/*
+ * QEMU Mixing engine
+ * 
+ * Copyright (c) 2004 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.
+ */
+
+/*
+ * Tusen tack till Mike Nordell
+ * dec++'ified by Dscho
+ */
+
+#ifdef SIGNED
+#define HALFT IN_MAX
+#define HALF IN_MAX
+#else
+#define HALFT ((IN_MAX)>>1)
+#define HALF HALFT
+#endif
+
+static int64_t inline glue(conv_,IN_T) (IN_T v)
+{
+#ifdef SIGNED
+    return (INT_MAX*(int64_t)v)/HALF;
+#else
+    return (INT_MAX*((int64_t)v-HALFT))/HALF;
+#endif
+}
+
+static IN_T inline glue(clip_,IN_T) (int64_t v)
+{
+    if (v >= INT_MAX)
+        return IN_MAX;
+    else if (v < -INT_MAX)
+        return IN_MIN;
+
+#ifdef SIGNED
+    return (IN_T) (v*HALF/INT_MAX);
+#else
+    return (IN_T) (v+INT_MAX/2)*HALF/INT_MAX;
+#endif
+}
+
+static void glue(glue(conv_,IN_T),_to_stereo) (void *dst, const void *src,
+                                               int samples)
+{
+    st_sample_t *out = (st_sample_t *) dst;
+    IN_T *in = (IN_T *) src;
+    while (samples--) {
+        out->l = glue(conv_,IN_T) (*in++);
+        out->r = glue(conv_,IN_T) (*in++);
+        out += 1;
+    }
+}
+
+static void glue(glue(conv_,IN_T),_to_mono) (void *dst, const void *src,
+                                             int samples)
+{
+    st_sample_t *out = (st_sample_t *) dst;
+    IN_T *in = (IN_T *) src;
+    while (samples--) {
+        out->l = glue(conv_,IN_T) (in[0]);
+        out->r = out->l;
+        out += 1;
+        in += 1;
+    }
+}
+
+static void glue(glue(clip_,IN_T),_from_stereo) (void *dst, const void *src,
+                                                 int samples)
+{
+    st_sample_t *in = (st_sample_t *) src;
+    IN_T *out = (IN_T *) dst;
+    while (samples--) {
+        *out++ = glue(clip_,IN_T) (in->l);
+        *out++ = glue(clip_,IN_T) (in->r);
+        in += 1;
+    }
+}
+
+static void glue(glue(clip_,IN_T),_from_mono) (void *dst, const void *src,
+                                               int samples)
+{
+    st_sample_t *in = (st_sample_t *) src;
+    IN_T *out = (IN_T *) dst;
+    while (samples--) {
+        *out++ = glue(clip_,IN_T) (in->l + in->r);
+        in += 1;
+    }
+}
+
+#undef HALF
+#undef HALFT
+
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
new file mode 100644
index 0000000..9fefaa3
--- /dev/null
+++ b/audio/ossaudio.c
@@ -0,0 +1,466 @@
+/*
+ * QEMU OSS audio output driver
+ * 
+ * Copyright (c) 2003-2004 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.
+ */
+
+/* Temporary kludge */
+#if defined __linux__ || (defined _BSD && !defined __APPLE__)
+#include <assert.h>
+#include "vl.h"
+
+#include <sys/mman.h>
+#include <sys/types.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+
+#define AUDIO_CAP "oss"
+#include "audio/audio.h"
+#include "audio/ossaudio.h"
+
+#define QC_OSS_FRAGSIZE "QEMU_OSS_FRAGSIZE"
+#define QC_OSS_NFRAGS   "QEMU_OSS_NFRAGS"
+#define QC_OSS_MMAP     "QEMU_OSS_MMAP"
+#define QC_OSS_DEV      "QEMU_OSS_DEV"
+
+#define errstr() strerror (errno)
+
+static struct {
+    int try_mmap;
+    int nfrags;
+    int fragsize;
+    const char *dspname;
+} conf = {
+    .try_mmap = 0,
+    .nfrags = 4,
+    .fragsize = 4096,
+    .dspname = "/dev/dsp"
+};
+
+struct oss_params {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    int nfrags;
+    int fragsize;
+};
+
+static int oss_hw_write (SWVoice *sw, void *buf, int len)
+{
+    return pcm_hw_write (sw, buf, len);
+}
+
+static int AUD_to_ossfmt (audfmt_e fmt)
+{
+    switch (fmt) {
+    case AUD_FMT_S8: return AFMT_S8;
+    case AUD_FMT_U8: return AFMT_U8;
+    case AUD_FMT_S16: return AFMT_S16_LE;
+    case AUD_FMT_U16: return AFMT_U16_LE;
+    default:
+        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
+        exit (EXIT_FAILURE);
+    }
+}
+
+static int oss_to_audfmt (int fmt)
+{
+    switch (fmt) {
+    case AFMT_S8: return AUD_FMT_S8;
+    case AFMT_U8: return AUD_FMT_U8;
+    case AFMT_S16_LE: return AUD_FMT_S16;
+    case AFMT_U16_LE: return AUD_FMT_U16;
+    default:
+        dolog ("Internal logic error: Unrecognized OSS audio format %d\n"
+               "Aborting\n",
+               fmt);
+        exit (EXIT_FAILURE);
+    }
+}
+
+#ifdef DEBUG_PCM
+static void oss_dump_pcm_info (struct oss_params *req, struct oss_params *obt)
+{
+    dolog ("parameter | requested value | obtained value\n");
+    dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
+    dolog ("channels  |      %10d |     %10d\n", req->nchannels, obt->nchannels);
+    dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
+    dolog ("nfrags    |      %10d |     %10d\n", req->nfrags, obt->nfrags);
+    dolog ("fragsize  |      %10d |     %10d\n", req->fragsize, obt->fragsize);
+}
+#endif
+
+static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
+{
+    int fd;
+    int mmmmssss;
+    audio_buf_info abinfo;
+    int fmt, freq, nchannels;
+    const char *dspname = conf.dspname;
+
+    fd = open (dspname, O_RDWR | O_NONBLOCK);
+    if (-1 == fd) {
+        dolog ("Could not initialize audio hardware. Failed to open `%s':\n"
+               "Reason:%s\n",
+               dspname,
+               errstr ());
+        return -1;
+    }
+
+    freq = req->freq;
+    nchannels = req->nchannels;
+    fmt = req->fmt;
+
+    if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
+        dolog ("Could not initialize audio hardware\n"
+               "Failed to set sample size\n"
+               "Reason: %s\n",
+               errstr ());
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) {
+        dolog ("Could not initialize audio hardware\n"
+               "Failed to set number of channels\n"
+               "Reason: %s\n",
+               errstr ());
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) {
+        dolog ("Could not initialize audio hardware\n"
+               "Failed to set frequency\n"
+               "Reason: %s\n",
+               errstr ());
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_NONBLOCK)) {
+        dolog ("Could not initialize audio hardware\n"
+               "Failed to set non-blocking mode\n"
+               "Reason: %s\n",
+               errstr ());
+        goto err;
+    }
+
+    mmmmssss = (req->nfrags << 16) | lsbindex (req->fragsize);
+    if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) {
+        dolog ("Could not initialize audio hardware\n"
+               "Failed to set buffer length (%d, %d)\n"
+               "Reason:%s\n",
+               conf.nfrags, conf.fragsize,
+               errstr ());
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &abinfo)) {
+        dolog ("Could not initialize audio hardware\n"
+               "Failed to get buffer length\n"
+               "Reason:%s\n",
+               errstr ());
+        goto err;
+    }
+
+    obt->fmt = fmt;
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->nfrags = abinfo.fragstotal;
+    obt->fragsize = abinfo.fragsize;
+    *pfd = fd;
+
+    if ((req->fmt != obt->fmt) ||
+        (req->nchannels != obt->nchannels) ||
+        (req->freq != obt->freq) ||
+        (req->fragsize != obt->fragsize) ||
+        (req->nfrags != obt->nfrags)) {
+#ifdef DEBUG_PCM
+        dolog ("Audio parameters mismatch\n");
+        oss_dump_pcm_info (req, obt);
+#endif
+    }
+
+#ifdef DEBUG_PCM
+    oss_dump_pcm_info (req, obt);
+#endif
+    return 0;
+
+err:
+    close (fd);
+    return -1;
+}
+
+static void oss_hw_run (HWVoice *hw)
+{
+    OSSVoice *oss = (OSSVoice *) hw;
+    int err, rpos, live, decr;
+    int samples;
+    uint8_t *dst;
+    st_sample_t *src;
+    struct audio_buf_info abinfo;
+    struct count_info cntinfo;
+
+    live = pcm_hw_get_live (hw);
+    if (live <= 0)
+        return;
+
+    if (oss->mmapped) {
+        int bytes;
+
+        err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
+        if (err < 0) {
+            dolog ("SNDCTL_DSP_GETOPTR failed\nReason: %s\n", errstr ());
+            return;
+        }
+
+        if (cntinfo.ptr == oss->old_optr) {
+            if (abs (hw->samples - live) < 64)
+                dolog ("overrun\n");
+            return;
+        }
+
+        if (cntinfo.ptr > oss->old_optr) {
+            bytes = cntinfo.ptr - oss->old_optr;
+        }
+        else {
+            bytes = hw->bufsize + cntinfo.ptr - oss->old_optr;
+        }
+
+        decr = audio_MIN (bytes >> hw->shift, live);
+    }
+    else {
+        err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
+        if (err < 0) {
+            dolog ("SNDCTL_DSP_GETOSPACE failed\nReason: %s\n", errstr ());
+            return;
+        }
+
+        decr = audio_MIN (abinfo.bytes >> hw->shift, live);
+        if (decr <= 0)
+            return;
+    }
+
+    samples = decr;
+    rpos = hw->rpos;
+    while (samples) {
+        int left_till_end_samples = hw->samples - rpos;
+        int convert_samples = audio_MIN (samples, left_till_end_samples);
+
+        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
+        dst = advance (oss->pcm_buf, rpos << hw->shift);
+
+        hw->clip (dst, src, convert_samples);
+        if (!oss->mmapped) {
+            int written;
+
+            written = write (oss->fd, dst, convert_samples << hw->shift);
+            /* XXX: follow errno recommendations ? */
+            if (written == -1) {
+                dolog ("Failed to write audio\nReason: %s\n", errstr ());
+                continue;
+            }
+
+            if (written != convert_samples << hw->shift) {
+                int wsamples = written >> hw->shift;
+                int wbytes = wsamples << hw->shift;
+                if (wbytes != written) {
+                    dolog ("Unaligned write %d, %d\n", wbytes, written);
+                }
+                memset (src, 0, wbytes);
+                decr -= samples;
+                rpos = (rpos + wsamples) % hw->samples;
+                break;
+            }
+        }
+        memset (src, 0, convert_samples * sizeof (st_sample_t));
+
+        rpos = (rpos + convert_samples) % hw->samples;
+        samples -= convert_samples;
+    }
+    if (oss->mmapped) {
+        oss->old_optr = cntinfo.ptr;
+    }
+
+    pcm_hw_dec_live (hw, decr);
+    hw->rpos = rpos;
+}
+
+static void oss_hw_fini (HWVoice *hw)
+{
+    int err;
+    OSSVoice *oss = (OSSVoice *) hw;
+
+    ldebug ("oss_hw_fini\n");
+    err = close (oss->fd);
+    if (err) {
+        dolog ("Failed to close OSS descriptor\nReason: %s\n", errstr ());
+    }
+    oss->fd = -1;
+
+    if (oss->pcm_buf) {
+        if (oss->mmapped) {
+            err = munmap (oss->pcm_buf, hw->bufsize);
+            if (err) {
+                dolog ("Failed to unmap OSS buffer\nReason: %s\n",
+                       errstr ());
+            }
+        }
+        else {
+            qemu_free (oss->pcm_buf);
+        }
+        oss->pcm_buf = NULL;
+    }
+}
+
+static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    OSSVoice *oss = (OSSVoice *) hw;
+    struct oss_params req, obt;
+
+    assert (!oss->fd);
+    req.fmt = AUD_to_ossfmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.fragsize = conf.fragsize;
+    req.nfrags = conf.nfrags;
+
+    if (oss_open (&req, &obt, &oss->fd))
+        return -1;
+
+    hw->freq = obt.freq;
+    hw->fmt = oss_to_audfmt (obt.fmt);
+    hw->nchannels = obt.nchannels;
+
+    oss->nfrags = obt.nfrags;
+    oss->fragsize = obt.fragsize;
+    hw->bufsize = obt.nfrags * obt.fragsize;
+
+    oss->mmapped = 0;
+    if (conf.try_mmap) {
+        oss->pcm_buf = mmap (0, hw->bufsize, PROT_READ | PROT_WRITE,
+                             MAP_SHARED, oss->fd, 0);
+        if (oss->pcm_buf == MAP_FAILED) {
+            dolog ("Failed to mmap OSS device\nReason: %s\n",
+                   errstr ());
+        }
+
+        for (;;) {
+            int err;
+            int trig = 0;
+            if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
+                       errstr ());
+                goto fail;
+            }
+
+            trig = PCM_ENABLE_OUTPUT;
+            if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                       "Reason: %s\n", errstr ());
+                goto fail;
+            }
+            oss->mmapped = 1;
+            break;
+
+        fail:
+            err = munmap (oss->pcm_buf, hw->bufsize);
+            if (err) {
+                dolog ("Failed to unmap OSS device\nReason: %s\n",
+                       errstr ());
+            }
+        }
+    }
+
+    if (!oss->mmapped) {
+        oss->pcm_buf = qemu_mallocz (hw->bufsize);
+        if (!oss->pcm_buf) {
+            close (oss->fd);
+            oss->fd = -1;
+            return -1;
+        }
+    }
+
+    return 0;
+}
+
+static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
+{
+    int trig;
+    OSSVoice *oss = (OSSVoice *) hw;
+
+    if (!oss->mmapped)
+        return 0;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        ldebug ("enabling voice\n");
+        pcm_hw_clear (hw, oss->pcm_buf, hw->samples);
+        trig = PCM_ENABLE_OUTPUT;
+        if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+            dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                   "Reason: %s\n", errstr ());
+            return -1;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        ldebug ("disabling voice\n");
+        trig = 0;
+        if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+            dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
+                   errstr ());
+            return -1;
+        }
+        break;
+    }
+    return 0;
+}
+
+static void *oss_audio_init (void)
+{
+    conf.fragsize = audio_get_conf_int (QC_OSS_FRAGSIZE, conf.fragsize);
+    conf.nfrags = audio_get_conf_int (QC_OSS_NFRAGS, conf.nfrags);
+    conf.try_mmap = audio_get_conf_int (QC_OSS_MMAP, conf.try_mmap);
+    conf.dspname = audio_get_conf_str (QC_OSS_DEV, conf.dspname);
+    return &conf;
+}
+
+static void oss_audio_fini (void *opaque)
+{
+}
+
+struct pcm_ops oss_pcm_ops = {
+    oss_hw_init,
+    oss_hw_fini,
+    oss_hw_run,
+    oss_hw_write,
+    oss_hw_ctl
+};
+
+struct audio_output_driver oss_output_driver = {
+    "oss",
+    oss_audio_init,
+    oss_audio_fini,
+    &oss_pcm_ops,
+    1,
+    INT_MAX,
+    sizeof (OSSVoice)
+};
+#endif
diff --git a/audio/ossaudio.h b/audio/ossaudio.h
new file mode 100644
index 0000000..f7d3ebd
--- /dev/null
+++ b/audio/ossaudio.h
@@ -0,0 +1,40 @@
+/*
+ * QEMU OSS audio output driver header
+ * 
+ * Copyright (c) 2003-2004 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.
+ */
+#ifndef QEMU_OSSAUDIO_H
+#define QEMU_OSSAUDIO_H
+
+typedef struct OSSVoice {
+    struct HWVoice hw;
+    void *pcm_buf;
+    int fd;
+    int nfrags;
+    int fragsize;
+    int mmapped;
+    int old_optr;
+} OSSVoice;
+
+extern struct pcm_ops oss_pcm_ops;
+extern struct audio_output_driver oss_output_driver;
+
+#endif  /* ossaudio.h */
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
new file mode 100644
index 0000000..4d75853
--- /dev/null
+++ b/audio/sdlaudio.c
@@ -0,0 +1,323 @@
+/*
+ * QEMU SDL audio output driver
+ * 
+ * Copyright (c) 2004 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/SDL.h>
+#include <SDL/SDL_thread.h>
+#include "vl.h"
+
+#define AUDIO_CAP "sdl"
+#include "audio/audio.h"
+#include "audio/sdlaudio.h"
+
+#define QC_SDL_SAMPLES "QEMU_SDL_SAMPLES"
+
+#define errstr() SDL_GetError ()
+
+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 sdl_hw_run (HWVoice *hw)
+{
+    (void) hw;
+}
+
+static int sdl_lock (SDLAudioState *s)
+{
+    if (SDL_LockMutex (s->mutex)) {
+        dolog ("SDL_LockMutex failed\nReason: %s\n", errstr ());
+        return -1;
+    }
+    return 0;
+}
+
+static int sdl_unlock (SDLAudioState *s)
+{
+    if (SDL_UnlockMutex (s->mutex)) {
+        dolog ("SDL_UnlockMutex failed\nReason: %s\n", errstr ());
+        return -1;
+    }
+    return 0;
+}
+
+static int sdl_post (SDLAudioState *s)
+{
+    if (SDL_SemPost (s->sem)) {
+        dolog ("SDL_SemPost failed\nReason: %s\n", errstr ());
+        return -1;
+    }
+    return 0;
+}
+
+static int sdl_wait (SDLAudioState *s)
+{
+    if (SDL_SemWait (s->sem)) {
+        dolog ("SDL_SemWait failed\nReason: %s\n", errstr ());
+        return -1;
+    }
+    return 0;
+}
+
+static int sdl_unlock_and_post (SDLAudioState *s)
+{
+    if (sdl_unlock (s))
+        return -1;
+
+    return sdl_post (s);
+}
+
+static int sdl_hw_write (SWVoice *sw, void *buf, int len)
+{
+    int ret;
+    SDLAudioState *s = &glob_sdl;
+    sdl_lock (s);
+    ret = pcm_hw_write (sw, buf, len);
+    sdl_unlock_and_post (s);
+    return ret;
+}
+
+static int AUD_to_sdlfmt (audfmt_e fmt, int *shift)
+{
+    *shift = 0;
+    switch (fmt) {
+    case AUD_FMT_S8: return AUDIO_S8;
+    case AUD_FMT_U8: 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\nAborting\n", fmt);
+        exit (EXIT_FAILURE);
+    }
+}
+
+static int sdl_to_audfmt (int fmt)
+{
+    switch (fmt) {
+    case AUDIO_S8: return AUD_FMT_S8;
+    case AUDIO_U8: return AUD_FMT_U8;
+    case AUDIO_S16LSB: return AUD_FMT_S16;
+    case AUDIO_U16LSB: return AUD_FMT_U16;
+    default:
+        dolog ("Internal logic error: Unrecognized SDL audio format %d\n"
+               "Aborting\n", fmt);
+        exit (EXIT_FAILURE);
+    }
+}
+
+static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
+{
+    int status;
+
+    status = SDL_OpenAudio (req, obt);
+    if (status) {
+        dolog ("SDL_OpenAudio failed\nReason: %s\n", errstr ());
+    }
+    return status;
+}
+
+static void sdl_close (SDLAudioState *s)
+{
+    if (s->initialized) {
+        sdl_lock (s);
+        s->exit = 1;
+        sdl_unlock_and_post (s);
+        SDL_PauseAudio (1);
+        SDL_CloseAudio ();
+        s->initialized = 0;
+    }
+}
+
+static void sdl_callback (void *opaque, Uint8 *buf, int len)
+{
+    SDLVoice *sdl = opaque;
+    SDLAudioState *s = &glob_sdl;
+    HWVoice *hw = &sdl->hw;
+    int samples = len >> hw->shift;
+
+    if (s->exit) {
+        return;
+    }
+
+    while (samples) {
+        int to_mix, live, decr;
+
+        /* dolog ("in callback samples=%d\n", samples); */
+        sdl_wait (s);
+        if (s->exit) {
+            return;
+        }
+
+        sdl_lock (s);
+        live = pcm_hw_get_live (hw);
+        if (live <= 0)
+            goto again;
+
+        /* dolog ("in callback live=%d\n", live); */
+        to_mix = audio_MIN (samples, 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);
+            memset (src, 0, chunk * sizeof (st_sample_t));
+            hw->rpos = (hw->rpos + chunk) % hw->samples;
+            to_mix -= chunk;
+            buf += chunk << hw->shift;
+        }
+        samples -= decr;
+        pcm_hw_dec_live (hw, decr);
+
+    again:
+        sdl_unlock (s);
+    }
+    /* dolog ("done len=%d\n", len); */
+}
+
+static void sdl_hw_fini (HWVoice *hw)
+{
+    ldebug ("sdl_hw_fini %d fixed=%d\n",
+             glob_sdl.initialized, audio_conf.fixed_format);
+    sdl_close (&glob_sdl);
+}
+
+static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    SDLVoice *sdl = (SDLVoice *) hw;
+    SDLAudioState *s = &glob_sdl;
+    SDL_AudioSpec req, obt;
+    int shift;
+
+    ldebug ("sdl_hw_init %d freq=%d fixed=%d\n",
+            s->initialized, freq, audio_conf.fixed_format);
+
+    if (nchannels != 2) {
+        dolog ("Bogus channel count %d\n", nchannels);
+        return -1;
+    }
+
+    req.freq = freq;
+    req.format = AUD_to_sdlfmt (fmt, &shift);
+    req.channels = nchannels;
+    req.samples = conf.nb_samples;
+    shift <<= nchannels == 2;
+
+    req.callback = sdl_callback;
+    req.userdata = sdl;
+
+    if (sdl_open (&req, &obt))
+        return -1;
+
+    hw->freq = obt.freq;
+    hw->fmt = sdl_to_audfmt (obt.format);
+    hw->nchannels = obt.channels;
+    hw->bufsize = obt.samples << shift;
+
+    s->initialized = 1;
+    s->exit = 0;
+    SDL_PauseAudio (0);
+    return 0;
+}
+
+static int sdl_hw_ctl (HWVoice *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;
+    conf.nb_samples = audio_get_conf_int (QC_SDL_SAMPLES, conf.nb_samples);
+
+    if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
+        dolog ("SDL failed to initialize audio subsystem\nReason: %s\n",
+               errstr ());
+        return NULL;
+    }
+
+    s->mutex = SDL_CreateMutex ();
+    if (!s->mutex) {
+        dolog ("Failed to create SDL mutex\nReason: %s\n", errstr ());
+        SDL_QuitSubSystem (SDL_INIT_AUDIO);
+        return NULL;
+    }
+
+    s->sem = SDL_CreateSemaphore (0);
+    if (!s->sem) {
+        dolog ("Failed to create SDL semaphore\nReason: %s\n", errstr ());
+        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);
+}
+
+struct pcm_ops sdl_pcm_ops = {
+    sdl_hw_init,
+    sdl_hw_fini,
+    sdl_hw_run,
+    sdl_hw_write,
+    sdl_hw_ctl
+};
+
+struct audio_output_driver sdl_output_driver = {
+    "sdl",
+    sdl_audio_init,
+    sdl_audio_fini,
+    &sdl_pcm_ops,
+    1,
+    1,
+    sizeof (SDLVoice)
+};
diff --git a/audio/sdlaudio.h b/audio/sdlaudio.h
new file mode 100644
index 0000000..380d0da
--- /dev/null
+++ b/audio/sdlaudio.h
@@ -0,0 +1,34 @@
+/*
+ * QEMU SDL audio output driver header
+ * 
+ * Copyright (c) 2004 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.
+ */
+#ifndef QEMU_SDLAUDIO_H
+#define QEMU_SDLAUDIO_H
+
+typedef struct SDLVoice {
+    struct HWVoice hw;
+} SDLVoice;
+
+extern struct pcm_ops sdl_pcm_ops;
+extern struct audio_output_driver sdl_output_driver;
+
+#endif  /* sdlaudio.h */
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
new file mode 100644
index 0000000..dee4a06
--- /dev/null
+++ b/audio/wavaudio.c
@@ -0,0 +1,200 @@
+/*
+ * QEMU WAV audio output driver
+ * 
+ * Copyright (c) 2004 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 "vl.h"
+
+#define AUDIO_CAP "wav"
+#include "audio/audio.h"
+#include "audio/wavaudio.h"
+
+static struct {
+    const char *wav_path;
+} conf = {
+    .wav_path = "qemu.wav"
+};
+
+static void wav_hw_run (HWVoice *hw)
+{
+    WAVVoice *wav = (WAVVoice *) hw;
+    int rpos, live, decr, samples;
+    uint8_t *dst;
+    st_sample_t *src;
+    int64_t now = qemu_get_clock (vm_clock);
+    int64_t ticks = now - wav->old_ticks;
+    int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+    wav->old_ticks = now;
+
+    if (bytes > INT_MAX)
+        samples = INT_MAX >> hw->shift;
+    else
+        samples = bytes >> hw->shift;
+
+    live = pcm_hw_get_live (hw);
+    if (live <= 0)
+        return;
+
+    decr = audio_MIN (live, samples);
+    samples = decr;
+    rpos = hw->rpos;
+    while (samples) {
+        int left_till_end_samples = hw->samples - rpos;
+        int convert_samples = audio_MIN (samples, left_till_end_samples);
+
+        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
+        dst = advance (wav->pcm_buf, rpos << hw->shift);
+
+        hw->clip (dst, src, convert_samples);
+        qemu_put_buffer (wav->f, dst, convert_samples << hw->shift);
+        memset (src, 0, convert_samples * sizeof (st_sample_t));
+
+        rpos = (rpos + convert_samples) % hw->samples;
+        samples -= convert_samples;
+        wav->total_samples += convert_samples;
+    }
+
+    pcm_hw_dec_live (hw, decr);
+    hw->rpos = rpos;
+}
+
+static int wav_hw_write (SWVoice *sw, void *buf, int len)
+{
+    return pcm_hw_write (sw, buf, len);
+}
+
+
+/* VICE code: Store number as little endian. */
+static void le_store (uint8_t *buf, uint32_t val, int len)
+{
+    int i;
+    for (i = 0; i < len; i++) {
+        buf[i] = (uint8_t) (val & 0xff);
+        val >>= 8;
+    }
+}
+
+static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    WAVVoice *wav = (WAVVoice *) hw;
+    int bits16 = 0, stereo = audio_state.fixed_channels == 2;
+    uint8_t hdr[] = {
+        0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56,
+        0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
+        0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
+        0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
+    };
+
+    switch (audio_state.fixed_fmt) {
+    case AUD_FMT_S8:
+    case AUD_FMT_U8:
+        break;
+
+    case AUD_FMT_S16:
+    case AUD_FMT_U16:
+        bits16 = 1;
+        break;
+    }
+
+    hdr[34] = bits16 ? 0x10 : 0x08;
+    hw->freq = 44100;
+    hw->nchannels = stereo ? 2 : 1;
+    hw->fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+    hw->bufsize = 4096;
+    wav->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!wav->pcm_buf)
+        return -1;
+
+    le_store (hdr + 22, hw->nchannels, 2);
+    le_store (hdr + 24, hw->freq, 4);
+    le_store (hdr + 28, hw->freq << (bits16 + stereo), 4);
+    le_store (hdr + 32, 1 << (bits16 + stereo), 2);
+
+    wav->f = fopen (conf.wav_path, "wb");
+    if (!wav->f) {
+        dolog ("failed to open wave file `%s'\nReason: %s\n",
+               conf.wav_path, strerror (errno));
+        return -1;
+    }
+
+    qemu_put_buffer (wav->f, hdr, sizeof (hdr));
+    return 0;
+}
+
+static void wav_hw_fini (HWVoice *hw)
+{
+    WAVVoice *wav = (WAVVoice *) hw;
+    int stereo = hw->nchannels == 2;
+    uint8_t rlen[4];
+    uint8_t dlen[4];
+    uint32_t rifflen = (wav->total_samples << stereo) + 36;
+    uint32_t datalen = wav->total_samples << stereo;
+
+    if (!wav->f || !hw->active)
+        return;
+
+    le_store (rlen, rifflen, 4);
+    le_store (dlen, datalen, 4);
+
+    qemu_fseek (wav->f, 4, SEEK_SET);
+    qemu_put_buffer (wav->f, rlen, 4);
+
+    qemu_fseek (wav->f, 32, SEEK_CUR);
+    qemu_put_buffer (wav->f, dlen, 4);
+
+    fclose (wav->f);
+    wav->f = NULL;
+}
+
+static int wav_hw_ctl (HWVoice *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static void *wav_audio_init (void)
+{
+    return &conf;
+}
+
+static void wav_audio_fini (void *opaque)
+{
+    ldebug ("wav_fini");
+}
+
+struct pcm_ops wav_pcm_ops = {
+    wav_hw_init,
+    wav_hw_fini,
+    wav_hw_run,
+    wav_hw_write,
+    wav_hw_ctl
+};
+
+struct audio_output_driver wav_output_driver = {
+    "wav",
+    wav_audio_init,
+    wav_audio_fini,
+    &wav_pcm_ops,
+    1,
+    1,
+    sizeof (WAVVoice)
+};
diff --git a/audio/wavaudio.h b/audio/wavaudio.h
new file mode 100644
index 0000000..0b6070b
--- /dev/null
+++ b/audio/wavaudio.h
@@ -0,0 +1,38 @@
+/*
+ * QEMU WAV audio output driver header
+ * 
+ * Copyright (c) 2004 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.
+ */
+#ifndef QEMU_WAVAUDIO_H
+#define QEMU_WAVAUDIO_H
+
+typedef struct WAVVoice {
+    struct HWVoice hw;
+    QEMUFile *f;
+    int64_t old_ticks;
+    void *pcm_buf;
+    int total_samples;
+} WAVVoice;
+
+extern struct pcm_ops wav_pcm_ops;
+extern struct audio_output_driver wav_output_driver;
+
+#endif  /* wavaudio.h */