Add qemu 2.4.0

Change-Id: Ic99cbad4b61f8b127b7dc74d04576c0bcbaaf4f5
Signed-off-by: Yang Zhang <yang.z.zhang@intel.com>

24 files changed, 10404 insertions, 0 deletions

diff --git a/qemu/audio/Makefile.objs b/qemu/audio/Makefile.objs
new file mode 100644
index 000000000..481d1aa30
--- /dev/null
+++ b/qemu/audio/Makefile.objs
@@ -0,0 +1,13 @@
+common-obj-y = audio.o noaudio.o wavaudio.o mixeng.o
+common-obj-$(CONFIG_SDL) += sdlaudio.o
+common-obj-$(CONFIG_OSS) += ossaudio.o
+common-obj-$(CONFIG_SPICE) += spiceaudio.o
+common-obj-$(CONFIG_COREAUDIO) += coreaudio.o
+common-obj-$(CONFIG_ALSA) += alsaaudio.o
+common-obj-$(CONFIG_DSOUND) += dsoundaudio.o
+common-obj-$(CONFIG_PA) += paaudio.o
+common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
+common-obj-$(CONFIG_AUDIO_WIN_INT) += audio_win_int.o
+common-obj-y += wavcapture.o
+
+sdlaudio.o-cflags := $(SDL_CFLAGS)
diff --git a/qemu/audio/alsaaudio.c b/qemu/audio/alsaaudio.c
new file mode 100644
index 000000000..6315b2d74
--- /dev/null
+++ b/qemu/audio/alsaaudio.c
@@ -0,0 +1,1227 @@
+/*
+ * QEMU ALSA audio driver
+ *
+ * Copyright (c) 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 <alsa/asoundlib.h>
+#include "qemu-common.h"
+#include "qemu/main-loop.h"
+#include "audio.h"
+#include "trace.h"
+
+#if QEMU_GNUC_PREREQ(4, 3)
+#pragma GCC diagnostic ignored "-Waddress"
+#endif
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAConf {
+    int size_in_usec_in;
+    int size_in_usec_out;
+    const char *pcm_name_in;
+    const char *pcm_name_out;
+    unsigned int buffer_size_in;
+    unsigned int period_size_in;
+    unsigned int buffer_size_out;
+    unsigned int period_size_out;
+    unsigned int threshold;
+
+    int buffer_size_in_overridden;
+    int period_size_in_overridden;
+
+    int buffer_size_out_overridden;
+    int period_size_out_overridden;
+} ALSAConf;
+
+struct pollhlp {
+    snd_pcm_t *handle;
+    struct pollfd *pfds;
+    ALSAConf *conf;
+    int count;
+    int mask;
+};
+
+typedef struct ALSAVoiceOut {
+    HWVoiceOut hw;
+    int wpos;
+    int pending;
+    void *pcm_buf;
+    snd_pcm_t *handle;
+    struct pollhlp pollhlp;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+    HWVoiceIn hw;
+    snd_pcm_t *handle;
+    void *pcm_buf;
+    struct pollhlp pollhlp;
+} ALSAVoiceIn;
+
+struct alsa_params_req {
+    int freq;
+    snd_pcm_format_t fmt;
+    int nchannels;
+    int size_in_usec;
+    int override_mask;
+    unsigned int buffer_size;
+    unsigned int period_size;
+};
+
+struct alsa_params_obt {
+    int freq;
+    audfmt_e fmt;
+    int endianness;
+    int nchannels;
+    snd_pcm_uframes_t samples;
+};
+
+static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, 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", snd_strerror (err));
+}
+
+static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
+    int err,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void alsa_fini_poll (struct pollhlp *hlp)
+{
+    int i;
+    struct pollfd *pfds = hlp->pfds;
+
+    if (pfds) {
+        for (i = 0; i < hlp->count; ++i) {
+            qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
+        }
+        g_free (pfds);
+    }
+    hlp->pfds = NULL;
+    hlp->count = 0;
+    hlp->handle = NULL;
+}
+
+static void alsa_anal_close1 (snd_pcm_t **handlep)
+{
+    int err = snd_pcm_close (*handlep);
+    if (err) {
+        alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
+    }
+    *handlep = NULL;
+}
+
+static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
+{
+    alsa_fini_poll (hlp);
+    alsa_anal_close1 (handlep);
+}
+
+static int alsa_recover (snd_pcm_t *handle)
+{
+    int err = snd_pcm_prepare (handle);
+    if (err < 0) {
+        alsa_logerr (err, "Failed to prepare handle %p\n", handle);
+        return -1;
+    }
+    return 0;
+}
+
+static int alsa_resume (snd_pcm_t *handle)
+{
+    int err = snd_pcm_resume (handle);
+    if (err < 0) {
+        alsa_logerr (err, "Failed to resume handle %p\n", handle);
+        return -1;
+    }
+    return 0;
+}
+
+static void alsa_poll_handler (void *opaque)
+{
+    int err, count;
+    snd_pcm_state_t state;
+    struct pollhlp *hlp = opaque;
+    unsigned short revents;
+
+    count = poll (hlp->pfds, hlp->count, 0);
+    if (count < 0) {
+        dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
+        return;
+    }
+
+    if (!count) {
+        return;
+    }
+
+    /* XXX: ALSA example uses initial count, not the one returned by
+       poll, correct? */
+    err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
+                                            hlp->count, &revents);
+    if (err < 0) {
+        alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
+        return;
+    }
+
+    if (!(revents & hlp->mask)) {
+        trace_alsa_revents(revents);
+        return;
+    }
+
+    state = snd_pcm_state (hlp->handle);
+    switch (state) {
+    case SND_PCM_STATE_SETUP:
+        alsa_recover (hlp->handle);
+        break;
+
+    case SND_PCM_STATE_XRUN:
+        alsa_recover (hlp->handle);
+        break;
+
+    case SND_PCM_STATE_SUSPENDED:
+        alsa_resume (hlp->handle);
+        break;
+
+    case SND_PCM_STATE_PREPARED:
+        audio_run ("alsa run (prepared)");
+        break;
+
+    case SND_PCM_STATE_RUNNING:
+        audio_run ("alsa run (running)");
+        break;
+
+    default:
+        dolog ("Unexpected state %d\n", state);
+    }
+}
+
+static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
+{
+    int i, count, err;
+    struct pollfd *pfds;
+
+    count = snd_pcm_poll_descriptors_count (handle);
+    if (count <= 0) {
+        dolog ("Could not initialize poll mode\n"
+               "Invalid number of poll descriptors %d\n", count);
+        return -1;
+    }
+
+    pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
+    if (!pfds) {
+        dolog ("Could not initialize poll mode\n");
+        return -1;
+    }
+
+    err = snd_pcm_poll_descriptors (handle, pfds, count);
+    if (err < 0) {
+        alsa_logerr (err, "Could not initialize poll mode\n"
+                     "Could not obtain poll descriptors\n");
+        g_free (pfds);
+        return -1;
+    }
+
+    for (i = 0; i < count; ++i) {
+        if (pfds[i].events & POLLIN) {
+            qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
+        }
+        if (pfds[i].events & POLLOUT) {
+            trace_alsa_pollout(i, pfds[i].fd);
+            qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
+        }
+        trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
+
+    }
+    hlp->pfds = pfds;
+    hlp->count = count;
+    hlp->handle = handle;
+    hlp->mask = mask;
+    return 0;
+}
+
+static int alsa_poll_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
+}
+
+static int alsa_poll_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
+}
+
+static int alsa_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+{
+    switch (fmt) {
+    case AUD_FMT_S8:
+        return SND_PCM_FORMAT_S8;
+
+    case AUD_FMT_U8:
+        return SND_PCM_FORMAT_U8;
+
+    case AUD_FMT_S16:
+        if (endianness) {
+            return SND_PCM_FORMAT_S16_BE;
+        }
+        else {
+            return SND_PCM_FORMAT_S16_LE;
+        }
+
+    case AUD_FMT_U16:
+        if (endianness) {
+            return SND_PCM_FORMAT_U16_BE;
+        }
+        else {
+            return SND_PCM_FORMAT_U16_LE;
+        }
+
+    case AUD_FMT_S32:
+        if (endianness) {
+            return SND_PCM_FORMAT_S32_BE;
+        }
+        else {
+            return SND_PCM_FORMAT_S32_LE;
+        }
+
+    case AUD_FMT_U32:
+        if (endianness) {
+            return SND_PCM_FORMAT_U32_BE;
+        }
+        else {
+            return SND_PCM_FORMAT_U32_LE;
+        }
+
+    default:
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+        abort ();
+#endif
+        return SND_PCM_FORMAT_U8;
+    }
+}
+
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+                           int *endianness)
+{
+    switch (alsafmt) {
+    case SND_PCM_FORMAT_S8:
+        *endianness = 0;
+        *fmt = AUD_FMT_S8;
+        break;
+
+    case SND_PCM_FORMAT_U8:
+        *endianness = 0;
+        *fmt = AUD_FMT_U8;
+        break;
+
+    case SND_PCM_FORMAT_S16_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case SND_PCM_FORMAT_U16_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case SND_PCM_FORMAT_S16_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case SND_PCM_FORMAT_U16_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case SND_PCM_FORMAT_S32_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_S32;
+        break;
+
+    case SND_PCM_FORMAT_U32_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_U32;
+        break;
+
+    case SND_PCM_FORMAT_S32_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_S32;
+        break;
+
+    case SND_PCM_FORMAT_U32_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_U32;
+        break;
+
+    default:
+        dolog ("Unrecognized audio format %d\n", alsafmt);
+        return -1;
+    }
+
+    return 0;
+}
+
+static void alsa_dump_info (struct alsa_params_req *req,
+                            struct alsa_params_obt *obt,
+                            snd_pcm_format_t obtfmt)
+{
+    dolog ("parameter | requested value | obtained value\n");
+    dolog ("format    |      %10d |     %10d\n", req->fmt, obtfmt);
+    dolog ("channels  |      %10d |     %10d\n",
+           req->nchannels, obt->nchannels);
+    dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
+    dolog ("============================================\n");
+    dolog ("requested: buffer size %d period size %d\n",
+           req->buffer_size, req->period_size);
+    dolog ("obtained: samples %ld\n", obt->samples);
+}
+
+static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
+{
+    int err;
+    snd_pcm_sw_params_t *sw_params;
+
+    snd_pcm_sw_params_alloca (&sw_params);
+
+    err = snd_pcm_sw_params_current (handle, sw_params);
+    if (err < 0) {
+        dolog ("Could not fully initialize DAC\n");
+        alsa_logerr (err, "Failed to get current software parameters\n");
+        return;
+    }
+
+    err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
+    if (err < 0) {
+        dolog ("Could not fully initialize DAC\n");
+        alsa_logerr (err, "Failed to set software threshold to %ld\n",
+                     threshold);
+        return;
+    }
+
+    err = snd_pcm_sw_params (handle, sw_params);
+    if (err < 0) {
+        dolog ("Could not fully initialize DAC\n");
+        alsa_logerr (err, "Failed to set software parameters\n");
+        return;
+    }
+}
+
+static int alsa_open (int in, struct alsa_params_req *req,
+                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
+                      ALSAConf *conf)
+{
+    snd_pcm_t *handle;
+    snd_pcm_hw_params_t *hw_params;
+    int err;
+    int size_in_usec;
+    unsigned int freq, nchannels;
+    const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+    snd_pcm_uframes_t obt_buffer_size;
+    const char *typ = in ? "ADC" : "DAC";
+    snd_pcm_format_t obtfmt;
+
+    freq = req->freq;
+    nchannels = req->nchannels;
+    size_in_usec = req->size_in_usec;
+
+    snd_pcm_hw_params_alloca (&hw_params);
+
+    err = snd_pcm_open (
+        &handle,
+        pcm_name,
+        in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
+        SND_PCM_NONBLOCK
+        );
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
+        return -1;
+    }
+
+    err = snd_pcm_hw_params_any (handle, hw_params);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
+        goto err;
+    }
+
+    err = snd_pcm_hw_params_set_access (
+        handle,
+        hw_params,
+        SND_PCM_ACCESS_RW_INTERLEAVED
+        );
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to set access type\n");
+        goto err;
+    }
+
+    err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
+    }
+
+    err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
+        goto err;
+    }
+
+    err = snd_pcm_hw_params_set_channels_near (
+        handle,
+        hw_params,
+        &nchannels
+        );
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
+                      req->nchannels);
+        goto err;
+    }
+
+    if (nchannels != 1 && nchannels != 2) {
+        alsa_logerr2 (err, typ,
+                      "Can not handle obtained number of channels %d\n",
+                      nchannels);
+        goto err;
+    }
+
+    if (req->buffer_size) {
+        unsigned long obt;
+
+        if (size_in_usec) {
+            int dir = 0;
+            unsigned int btime = req->buffer_size;
+
+            err = snd_pcm_hw_params_set_buffer_time_near (
+                handle,
+                hw_params,
+                &btime,
+                &dir
+                );
+            obt = btime;
+        }
+        else {
+            snd_pcm_uframes_t bsize = req->buffer_size;
+
+            err = snd_pcm_hw_params_set_buffer_size_near (
+                handle,
+                hw_params,
+                &bsize
+                );
+            obt = bsize;
+        }
+        if (err < 0) {
+            alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
+                          size_in_usec ? "time" : "size", req->buffer_size);
+            goto err;
+        }
+
+        if ((req->override_mask & 2) && (obt - req->buffer_size))
+            dolog ("Requested buffer %s %u was rejected, using %lu\n",
+                   size_in_usec ? "time" : "size", req->buffer_size, obt);
+    }
+
+    if (req->period_size) {
+        unsigned long obt;
+
+        if (size_in_usec) {
+            int dir = 0;
+            unsigned int ptime = req->period_size;
+
+            err = snd_pcm_hw_params_set_period_time_near (
+                handle,
+                hw_params,
+                &ptime,
+                &dir
+                );
+            obt = ptime;
+        }
+        else {
+            int dir = 0;
+            snd_pcm_uframes_t psize = req->period_size;
+
+            err = snd_pcm_hw_params_set_period_size_near (
+                handle,
+                hw_params,
+                &psize,
+                &dir
+                );
+            obt = psize;
+        }
+
+        if (err < 0) {
+            alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
+                          size_in_usec ? "time" : "size", req->period_size);
+            goto err;
+        }
+
+        if (((req->override_mask & 1) && (obt - req->period_size)))
+            dolog ("Requested period %s %u was rejected, using %lu\n",
+                   size_in_usec ? "time" : "size", req->period_size, obt);
+    }
+
+    err = snd_pcm_hw_params (handle, hw_params);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
+        goto err;
+    }
+
+    err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to get buffer size\n");
+        goto err;
+    }
+
+    err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Failed to get format\n");
+        goto err;
+    }
+
+    if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
+        dolog ("Invalid format was returned %d\n", obtfmt);
+        goto err;
+    }
+
+    err = snd_pcm_prepare (handle);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
+        goto err;
+    }
+
+    if (!in && conf->threshold) {
+        snd_pcm_uframes_t threshold;
+        int bytes_per_sec;
+
+        bytes_per_sec = freq << (nchannels == 2);
+
+        switch (obt->fmt) {
+        case AUD_FMT_S8:
+        case AUD_FMT_U8:
+            break;
+
+        case AUD_FMT_S16:
+        case AUD_FMT_U16:
+            bytes_per_sec <<= 1;
+            break;
+
+        case AUD_FMT_S32:
+        case AUD_FMT_U32:
+            bytes_per_sec <<= 2;
+            break;
+        }
+
+        threshold = (conf->threshold * bytes_per_sec) / 1000;
+        alsa_set_threshold (handle, threshold);
+    }
+
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->samples = obt_buffer_size;
+
+    *handlep = handle;
+
+    if (obtfmt != req->fmt ||
+         obt->nchannels != req->nchannels ||
+         obt->freq != req->freq) {
+        dolog ("Audio parameters for %s\n", typ);
+        alsa_dump_info (req, obt, obtfmt);
+    }
+
+#ifdef DEBUG
+    alsa_dump_info (req, obt, obtfmt);
+#endif
+    return 0;
+
+ err:
+    alsa_anal_close1 (&handle);
+    return -1;
+}
+
+static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
+{
+    snd_pcm_sframes_t avail;
+
+    avail = snd_pcm_avail_update (handle);
+    if (avail < 0) {
+        if (avail == -EPIPE) {
+            if (!alsa_recover (handle)) {
+                avail = snd_pcm_avail_update (handle);
+            }
+        }
+
+        if (avail < 0) {
+            alsa_logerr (avail,
+                         "Could not obtain number of available frames\n");
+            return -1;
+        }
+    }
+
+    return avail;
+}
+
+static void alsa_write_pending (ALSAVoiceOut *alsa)
+{
+    HWVoiceOut *hw = &alsa->hw;
+
+    while (alsa->pending) {
+        int left_till_end_samples = hw->samples - alsa->wpos;
+        int len = audio_MIN (alsa->pending, left_till_end_samples);
+        char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
+
+        while (len) {
+            snd_pcm_sframes_t written;
+
+            written = snd_pcm_writei (alsa->handle, src, len);
+
+            if (written <= 0) {
+                switch (written) {
+                case 0:
+                    trace_alsa_wrote_zero(len);
+                    return;
+
+                case -EPIPE:
+                    if (alsa_recover (alsa->handle)) {
+                        alsa_logerr (written, "Failed to write %d frames\n",
+                                     len);
+                        return;
+                    }
+                    trace_alsa_xrun_out();
+                    continue;
+
+                case -ESTRPIPE:
+                    /* stream is suspended and waiting for an
+                       application recovery */
+                    if (alsa_resume (alsa->handle)) {
+                        alsa_logerr (written, "Failed to write %d frames\n",
+                                     len);
+                        return;
+                    }
+                    trace_alsa_resume_out();
+                    continue;
+
+                case -EAGAIN:
+                    return;
+
+                default:
+                    alsa_logerr (written, "Failed to write %d frames from %p\n",
+                                 len, src);
+                    return;
+                }
+            }
+
+            alsa->wpos = (alsa->wpos + written) % hw->samples;
+            alsa->pending -= written;
+            len -= written;
+        }
+    }
+}
+
+static int alsa_run_out (HWVoiceOut *hw, int live)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    int decr;
+    snd_pcm_sframes_t avail;
+
+    avail = alsa_get_avail (alsa->handle);
+    if (avail < 0) {
+        dolog ("Could not get number of available playback frames\n");
+        return 0;
+    }
+
+    decr = audio_MIN (live, avail);
+    decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
+    alsa->pending += decr;
+    alsa_write_pending (alsa);
+    return decr;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    ldebug ("alsa_fini\n");
+    alsa_anal_close (&alsa->handle, &alsa->pollhlp);
+
+    g_free(alsa->pcm_buf);
+    alsa->pcm_buf = NULL;
+}
+
+static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
+                         void *drv_opaque)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    snd_pcm_t *handle;
+    struct audsettings obt_as;
+    ALSAConf *conf = drv_opaque;
+
+    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+    req.freq = as->freq;
+    req.nchannels = as->nchannels;
+    req.period_size = conf->period_size_out;
+    req.buffer_size = conf->buffer_size_out;
+    req.size_in_usec = conf->size_in_usec_out;
+    req.override_mask =
+        (conf->period_size_out_overridden ? 1 : 0) |
+        (conf->buffer_size_out_overridden ? 2 : 0);
+
+    if (alsa_open (0, &req, &obt, &handle, conf)) {
+        return -1;
+    }
+
+    obt_as.freq = obt.freq;
+    obt_as.nchannels = obt.nchannels;
+    obt_as.fmt = obt.fmt;
+    obt_as.endianness = obt.endianness;
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    hw->samples = obt.samples;
+
+    alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
+    if (!alsa->pcm_buf) {
+        dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
+               hw->samples, 1 << hw->info.shift);
+        alsa_anal_close1 (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    alsa->pollhlp.conf = conf;
+    return 0;
+}
+
+#define VOICE_CTL_PAUSE 0
+#define VOICE_CTL_PREPARE 1
+#define VOICE_CTL_START 2
+
+static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
+{
+    int err;
+
+    if (ctl == VOICE_CTL_PAUSE) {
+        err = snd_pcm_drop (handle);
+        if (err < 0) {
+            alsa_logerr (err, "Could not stop %s\n", typ);
+            return -1;
+        }
+    }
+    else {
+        err = snd_pcm_prepare (handle);
+        if (err < 0) {
+            alsa_logerr (err, "Could not prepare handle for %s\n", typ);
+            return -1;
+        }
+        if (ctl == VOICE_CTL_START) {
+            err = snd_pcm_start(handle);
+            if (err < 0) {
+                alsa_logerr (err, "Could not start handle for %s\n", typ);
+                return -1;
+            }
+        }
+    }
+
+    return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        {
+            va_list ap;
+            int poll_mode;
+
+            va_start (ap, cmd);
+            poll_mode = va_arg (ap, int);
+            va_end (ap);
+
+            ldebug ("enabling voice\n");
+            if (poll_mode && alsa_poll_out (hw)) {
+                poll_mode = 0;
+            }
+            hw->poll_mode = poll_mode;
+            return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
+        }
+
+    case VOICE_DISABLE:
+        ldebug ("disabling voice\n");
+        if (hw->poll_mode) {
+            hw->poll_mode = 0;
+            alsa_fini_poll (&alsa->pollhlp);
+        }
+        return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
+    }
+
+    return -1;
+}
+
+static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    snd_pcm_t *handle;
+    struct audsettings obt_as;
+    ALSAConf *conf = drv_opaque;
+
+    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+    req.freq = as->freq;
+    req.nchannels = as->nchannels;
+    req.period_size = conf->period_size_in;
+    req.buffer_size = conf->buffer_size_in;
+    req.size_in_usec = conf->size_in_usec_in;
+    req.override_mask =
+        (conf->period_size_in_overridden ? 1 : 0) |
+        (conf->buffer_size_in_overridden ? 2 : 0);
+
+    if (alsa_open (1, &req, &obt, &handle, conf)) {
+        return -1;
+    }
+
+    obt_as.freq = obt.freq;
+    obt_as.nchannels = obt.nchannels;
+    obt_as.fmt = obt.fmt;
+    obt_as.endianness = obt.endianness;
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    hw->samples = obt.samples;
+
+    alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    if (!alsa->pcm_buf) {
+        dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
+               hw->samples, 1 << hw->info.shift);
+        alsa_anal_close1 (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    alsa->pollhlp.conf = conf;
+    return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    alsa_anal_close (&alsa->handle, &alsa->pollhlp);
+
+    g_free(alsa->pcm_buf);
+    alsa->pcm_buf = NULL;
+}
+
+static int alsa_run_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    int hwshift = hw->info.shift;
+    int i;
+    int live = audio_pcm_hw_get_live_in (hw);
+    int dead = hw->samples - live;
+    int decr;
+    struct {
+        int add;
+        int len;
+    } bufs[2] = {
+        { .add = hw->wpos, .len = 0 },
+        { .add = 0,        .len = 0 }
+    };
+    snd_pcm_sframes_t avail;
+    snd_pcm_uframes_t read_samples = 0;
+
+    if (!dead) {
+        return 0;
+    }
+
+    avail = alsa_get_avail (alsa->handle);
+    if (avail < 0) {
+        dolog ("Could not get number of captured frames\n");
+        return 0;
+    }
+
+    if (!avail) {
+        snd_pcm_state_t state;
+
+        state = snd_pcm_state (alsa->handle);
+        switch (state) {
+        case SND_PCM_STATE_PREPARED:
+            avail = hw->samples;
+            break;
+        case SND_PCM_STATE_SUSPENDED:
+            /* stream is suspended and waiting for an application recovery */
+            if (alsa_resume (alsa->handle)) {
+                dolog ("Failed to resume suspended input stream\n");
+                return 0;
+            }
+            trace_alsa_resume_in();
+            break;
+        default:
+            trace_alsa_no_frames(state);
+            return 0;
+        }
+    }
+
+    decr = audio_MIN (dead, avail);
+    if (!decr) {
+        return 0;
+    }
+
+    if (hw->wpos + decr > hw->samples) {
+        bufs[0].len = (hw->samples - hw->wpos);
+        bufs[1].len = (decr - (hw->samples - hw->wpos));
+    }
+    else {
+        bufs[0].len = decr;
+    }
+
+    for (i = 0; i < 2; ++i) {
+        void *src;
+        struct st_sample *dst;
+        snd_pcm_sframes_t nread;
+        snd_pcm_uframes_t len;
+
+        len = bufs[i].len;
+
+        src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
+        dst = hw->conv_buf + bufs[i].add;
+
+        while (len) {
+            nread = snd_pcm_readi (alsa->handle, src, len);
+
+            if (nread <= 0) {
+                switch (nread) {
+                case 0:
+                    trace_alsa_read_zero(len);
+                    goto exit;
+
+                case -EPIPE:
+                    if (alsa_recover (alsa->handle)) {
+                        alsa_logerr (nread, "Failed to read %ld frames\n", len);
+                        goto exit;
+                    }
+                    trace_alsa_xrun_in();
+                    continue;
+
+                case -EAGAIN:
+                    goto exit;
+
+                default:
+                    alsa_logerr (
+                        nread,
+                        "Failed to read %ld frames from %p\n",
+                        len,
+                        src
+                        );
+                    goto exit;
+                }
+            }
+
+            hw->conv (dst, src, nread);
+
+            src = advance (src, nread << hwshift);
+            dst += nread;
+
+            read_samples += nread;
+            len -= nread;
+        }
+    }
+
+ exit:
+    hw->wpos = (hw->wpos + read_samples) % hw->samples;
+    return read_samples;
+}
+
+static int alsa_read (SWVoiceIn *sw, void *buf, int size)
+{
+    return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        {
+            va_list ap;
+            int poll_mode;
+
+            va_start (ap, cmd);
+            poll_mode = va_arg (ap, int);
+            va_end (ap);
+
+            ldebug ("enabling voice\n");
+            if (poll_mode && alsa_poll_in (hw)) {
+                poll_mode = 0;
+            }
+            hw->poll_mode = poll_mode;
+
+            return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
+        }
+
+    case VOICE_DISABLE:
+        ldebug ("disabling voice\n");
+        if (hw->poll_mode) {
+            hw->poll_mode = 0;
+            alsa_fini_poll (&alsa->pollhlp);
+        }
+        return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
+    }
+
+    return -1;
+}
+
+static ALSAConf glob_conf = {
+    .buffer_size_out = 4096,
+    .period_size_out = 1024,
+    .pcm_name_out = "default",
+    .pcm_name_in = "default",
+};
+
+static void *alsa_audio_init (void)
+{
+    ALSAConf *conf = g_malloc(sizeof(ALSAConf));
+    *conf = glob_conf;
+    return conf;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+    g_free(opaque);
+}
+
+static struct audio_option alsa_options[] = {
+    {
+        .name        = "DAC_SIZE_IN_USEC",
+        .tag         = AUD_OPT_BOOL,
+        .valp        = &glob_conf.size_in_usec_out,
+        .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
+    },
+    {
+        .name        = "DAC_PERIOD_SIZE",
+        .tag         = AUD_OPT_INT,
+        .valp        = &glob_conf.period_size_out,
+        .descr       = "DAC period size (0 to go with system default)",
+        .overriddenp = &glob_conf.period_size_out_overridden
+    },
+    {
+        .name        = "DAC_BUFFER_SIZE",
+        .tag         = AUD_OPT_INT,
+        .valp        = &glob_conf.buffer_size_out,
+        .descr       = "DAC buffer size (0 to go with system default)",
+        .overriddenp = &glob_conf.buffer_size_out_overridden
+    },
+    {
+        .name        = "ADC_SIZE_IN_USEC",
+        .tag         = AUD_OPT_BOOL,
+        .valp        = &glob_conf.size_in_usec_in,
+        .descr       =
+        "ADC period/buffer size in microseconds (otherwise in frames)"
+    },
+    {
+        .name        = "ADC_PERIOD_SIZE",
+        .tag         = AUD_OPT_INT,
+        .valp        = &glob_conf.period_size_in,
+        .descr       = "ADC period size (0 to go with system default)",
+        .overriddenp = &glob_conf.period_size_in_overridden
+    },
+    {
+        .name        = "ADC_BUFFER_SIZE",
+        .tag         = AUD_OPT_INT,
+        .valp        = &glob_conf.buffer_size_in,
+        .descr       = "ADC buffer size (0 to go with system default)",
+        .overriddenp = &glob_conf.buffer_size_in_overridden
+    },
+    {
+        .name        = "THRESHOLD",
+        .tag         = AUD_OPT_INT,
+        .valp        = &glob_conf.threshold,
+        .descr       = "(undocumented)"
+    },
+    {
+        .name        = "DAC_DEV",
+        .tag         = AUD_OPT_STR,
+        .valp        = &glob_conf.pcm_name_out,
+        .descr       = "DAC device name (for instance dmix)"
+    },
+    {
+        .name        = "ADC_DEV",
+        .tag         = AUD_OPT_STR,
+        .valp        = &glob_conf.pcm_name_in,
+        .descr       = "ADC device name"
+    },
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+    .init_out = alsa_init_out,
+    .fini_out = alsa_fini_out,
+    .run_out  = alsa_run_out,
+    .write    = alsa_write,
+    .ctl_out  = alsa_ctl_out,
+
+    .init_in  = alsa_init_in,
+    .fini_in  = alsa_fini_in,
+    .run_in   = alsa_run_in,
+    .read     = alsa_read,
+    .ctl_in   = alsa_ctl_in,
+};
+
+struct audio_driver alsa_audio_driver = {
+    .name           = "alsa",
+    .descr          = "ALSA http://www.alsa-project.org",
+    .options        = alsa_options,
+    .init           = alsa_audio_init,
+    .fini           = alsa_audio_fini,
+    .pcm_ops        = &alsa_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
+    .voice_size_out = sizeof (ALSAVoiceOut),
+    .voice_size_in  = sizeof (ALSAVoiceIn)
+};
diff --git a/qemu/audio/audio.c b/qemu/audio/audio.c
new file mode 100644
index 000000000..5be4b15fc
--- /dev/null
+++ b/qemu/audio/audio.c
@@ -0,0 +1,2063 @@
+/*
+ * QEMU Audio subsystem
+ *
+ * Copyright (c) 2003-2005 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 "hw/hw.h"
+#include "audio.h"
+#include "monitor/monitor.h"
+#include "qemu/timer.h"
+#include "sysemu/sysemu.h"
+
+#define AUDIO_CAP "audio"
+#include "audio_int.h"
+
+/* #define DEBUG_LIVE */
+/* #define DEBUG_OUT */
+/* #define DEBUG_CAPTURE */
+/* #define DEBUG_POLL */
+
+#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
+
+
+/* Order of CONFIG_AUDIO_DRIVERS is import.
+   The 1st one is the one used by default, that is the reason
+    that we generate the list.
+*/
+static struct audio_driver *drvtab[] = {
+#ifdef CONFIG_SPICE
+    &spice_audio_driver,
+#endif
+    CONFIG_AUDIO_DRIVERS
+    &no_audio_driver,
+    &wav_audio_driver
+};
+
+struct fixed_settings {
+    int enabled;
+    int nb_voices;
+    int greedy;
+    struct audsettings settings;
+};
+
+static struct {
+    struct fixed_settings fixed_out;
+    struct fixed_settings fixed_in;
+    union {
+        int hertz;
+        int64_t ticks;
+    } period;
+    int try_poll_in;
+    int try_poll_out;
+} conf = {
+    .fixed_out = { /* DAC fixed settings */
+        .enabled = 1,
+        .nb_voices = 1,
+        .greedy = 1,
+        .settings = {
+            .freq = 44100,
+            .nchannels = 2,
+            .fmt = AUD_FMT_S16,
+            .endianness =  AUDIO_HOST_ENDIANNESS,
+        }
+    },
+
+    .fixed_in = { /* ADC fixed settings */
+        .enabled = 1,
+        .nb_voices = 1,
+        .greedy = 1,
+        .settings = {
+            .freq = 44100,
+            .nchannels = 2,
+            .fmt = AUD_FMT_S16,
+            .endianness = AUDIO_HOST_ENDIANNESS,
+        }
+    },
+
+    .period = { .hertz = 100 },
+    .try_poll_in = 1,
+    .try_poll_out = 1,
+};
+
+static AudioState glob_audio_state;
+
+const struct mixeng_volume nominal_volume = {
+    .mute = 0,
+#ifdef FLOAT_MIXENG
+    .r = 1.0,
+    .l = 1.0,
+#else
+    .r = 1ULL << 32,
+    .l = 1ULL << 32,
+#endif
+};
+
+#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
+#error No its not
+#else
+static void audio_print_options (const char *prefix,
+                                 struct audio_option *opt);
+
+int audio_bug (const char *funcname, int cond)
+{
+    if (cond) {
+        static int shown;
+
+        AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
+        if (!shown) {
+            struct audio_driver *d;
+
+            shown = 1;
+            AUD_log (NULL, "Save all your work and restart without audio\n");
+            AUD_log (NULL, "Please send bug report to av1474@comtv.ru\n");
+            AUD_log (NULL, "I am sorry\n");
+            d = glob_audio_state.drv;
+            if (d) {
+                audio_print_options (d->name, d->options);
+            }
+        }
+        AUD_log (NULL, "Context:\n");
+
+#if defined AUDIO_BREAKPOINT_ON_BUG
+#  if defined HOST_I386
+#    if defined __GNUC__
+        __asm__ ("int3");
+#    elif defined _MSC_VER
+        _asm _emit 0xcc;
+#    else
+        abort ();
+#    endif
+#  else
+        abort ();
+#  endif
+#endif
+    }
+
+    return cond;
+}
+#endif
+
+static inline int audio_bits_to_index (int bits)
+{
+    switch (bits) {
+    case 8:
+        return 0;
+
+    case 16:
+        return 1;
+
+    case 32:
+        return 2;
+
+    default:
+        audio_bug ("bits_to_index", 1);
+        AUD_log (NULL, "invalid bits %d\n", bits);
+        return 0;
+    }
+}
+
+void *audio_calloc (const char *funcname, int nmemb, size_t size)
+{
+    int cond;
+    size_t len;
+
+    len = nmemb * size;
+    cond = !nmemb || !size;
+    cond |= nmemb < 0;
+    cond |= len < size;
+
+    if (audio_bug ("audio_calloc", cond)) {
+        AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
+                 funcname);
+        AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
+        return NULL;
+    }
+
+    return g_malloc0 (len);
+}
+
+static char *audio_alloc_prefix (const char *s)
+{
+    const char qemu_prefix[] = "QEMU_";
+    size_t len, i;
+    char *r, *u;
+
+    if (!s) {
+        return NULL;
+    }
+
+    len = strlen (s);
+    r = g_malloc (len + sizeof (qemu_prefix));
+
+    u = r + sizeof (qemu_prefix) - 1;
+
+    pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
+    pstrcat (r, len + sizeof (qemu_prefix), s);
+
+    for (i = 0; i < len; ++i) {
+        u[i] = qemu_toupper(u[i]);
+    }
+
+    return r;
+}
+
+static const char *audio_audfmt_to_string (audfmt_e fmt)
+{
+    switch (fmt) {
+    case AUD_FMT_U8:
+        return "U8";
+
+    case AUD_FMT_U16:
+        return "U16";
+
+    case AUD_FMT_S8:
+        return "S8";
+
+    case AUD_FMT_S16:
+        return "S16";
+
+    case AUD_FMT_U32:
+        return "U32";
+
+    case AUD_FMT_S32:
+        return "S32";
+    }
+
+    dolog ("Bogus audfmt %d returning S16\n", fmt);
+    return "S16";
+}
+
+static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+                                        int *defaultp)
+{
+    if (!strcasecmp (s, "u8")) {
+        *defaultp = 0;
+        return AUD_FMT_U8;
+    }
+    else if (!strcasecmp (s, "u16")) {
+        *defaultp = 0;
+        return AUD_FMT_U16;
+    }
+    else if (!strcasecmp (s, "u32")) {
+        *defaultp = 0;
+        return AUD_FMT_U32;
+    }
+    else if (!strcasecmp (s, "s8")) {
+        *defaultp = 0;
+        return AUD_FMT_S8;
+    }
+    else if (!strcasecmp (s, "s16")) {
+        *defaultp = 0;
+        return AUD_FMT_S16;
+    }
+    else if (!strcasecmp (s, "s32")) {
+        *defaultp = 0;
+        return AUD_FMT_S32;
+    }
+    else {
+        dolog ("Bogus audio format `%s' using %s\n",
+               s, audio_audfmt_to_string (defval));
+        *defaultp = 1;
+        return defval;
+    }
+}
+
+static audfmt_e audio_get_conf_fmt (const char *envname,
+                                    audfmt_e defval,
+                                    int *defaultp)
+{
+    const char *var = getenv (envname);
+    if (!var) {
+        *defaultp = 1;
+        return defval;
+    }
+    return audio_string_to_audfmt (var, defval, defaultp);
+}
+
+static int audio_get_conf_int (const char *key, int defval, int *defaultp)
+{
+    int val;
+    char *strval;
+
+    strval = getenv (key);
+    if (strval) {
+        *defaultp = 0;
+        val = atoi (strval);
+        return val;
+    }
+    else {
+        *defaultp = 1;
+        return defval;
+    }
+}
+
+static const char *audio_get_conf_str (const char *key,
+                                       const char *defval,
+                                       int *defaultp)
+{
+    const char *val = getenv (key);
+    if (!val) {
+        *defaultp = 1;
+        return defval;
+    }
+    else {
+        *defaultp = 0;
+        return val;
+    }
+}
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap)
+{
+    if (cap) {
+        fprintf(stderr, "%s: ", cap);
+    }
+
+    vfprintf(stderr, fmt, ap);
+}
+
+void AUD_log (const char *cap, const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (cap, fmt, ap);
+    va_end (ap);
+}
+
+static void audio_print_options (const char *prefix,
+                                 struct audio_option *opt)
+{
+    char *uprefix;
+
+    if (!prefix) {
+        dolog ("No prefix specified\n");
+        return;
+    }
+
+    if (!opt) {
+        dolog ("No options\n");
+        return;
+    }
+
+    uprefix = audio_alloc_prefix (prefix);
+
+    for (; opt->name; opt++) {
+        const char *state = "default";
+        printf ("  %s_%s: ", uprefix, opt->name);
+
+        if (opt->overriddenp && *opt->overriddenp) {
+            state = "current";
+        }
+
+        switch (opt->tag) {
+        case AUD_OPT_BOOL:
+            {
+                int *intp = opt->valp;
+                printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
+            }
+            break;
+
+        case AUD_OPT_INT:
+            {
+                int *intp = opt->valp;
+                printf ("integer, %s = %d\n", state, *intp);
+            }
+            break;
+
+        case AUD_OPT_FMT:
+            {
+                audfmt_e *fmtp = opt->valp;
+                printf (
+                    "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
+                    state,
+                    audio_audfmt_to_string (*fmtp)
+                    );
+            }
+            break;
+
+        case AUD_OPT_STR:
+            {
+                const char **strp = opt->valp;
+                printf ("string, %s = %s\n",
+                        state,
+                        *strp ? *strp : "(not set)");
+            }
+            break;
+
+        default:
+            printf ("???\n");
+            dolog ("Bad value tag for option %s_%s %d\n",
+                   uprefix, opt->name, opt->tag);
+            break;
+        }
+        printf ("    %s\n", opt->descr);
+    }
+
+    g_free (uprefix);
+}
+
+static void audio_process_options (const char *prefix,
+                                   struct audio_option *opt)
+{
+    char *optname;
+    const char qemu_prefix[] = "QEMU_";
+    size_t preflen, optlen;
+
+    if (audio_bug (AUDIO_FUNC, !prefix)) {
+        dolog ("prefix = NULL\n");
+        return;
+    }
+
+    if (audio_bug (AUDIO_FUNC, !opt)) {
+        dolog ("opt = NULL\n");
+        return;
+    }
+
+    preflen = strlen (prefix);
+
+    for (; opt->name; opt++) {
+        size_t len, i;
+        int def;
+
+        if (!opt->valp) {
+            dolog ("Option value pointer for `%s' is not set\n",
+                   opt->name);
+            continue;
+        }
+
+        len = strlen (opt->name);
+        /* len of opt->name + len of prefix + size of qemu_prefix
+         * (includes trailing zero) + zero + underscore (on behalf of
+         * sizeof) */
+        optlen = len + preflen + sizeof (qemu_prefix) + 1;
+        optname = g_malloc (optlen);
+
+        pstrcpy (optname, optlen, qemu_prefix);
+
+        /* copy while upper-casing, including trailing zero */
+        for (i = 0; i <= preflen; ++i) {
+            optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
+        }
+        pstrcat (optname, optlen, "_");
+        pstrcat (optname, optlen, opt->name);
+
+        def = 1;
+        switch (opt->tag) {
+        case AUD_OPT_BOOL:
+        case AUD_OPT_INT:
+            {
+                int *intp = opt->valp;
+                *intp = audio_get_conf_int (optname, *intp, &def);
+            }
+            break;
+
+        case AUD_OPT_FMT:
+            {
+                audfmt_e *fmtp = opt->valp;
+                *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
+            }
+            break;
+
+        case AUD_OPT_STR:
+            {
+                const char **strp = opt->valp;
+                *strp = audio_get_conf_str (optname, *strp, &def);
+            }
+            break;
+
+        default:
+            dolog ("Bad value tag for option `%s' - %d\n",
+                   optname, opt->tag);
+            break;
+        }
+
+        if (!opt->overriddenp) {
+            opt->overriddenp = &opt->overridden;
+        }
+        *opt->overriddenp = !def;
+        g_free (optname);
+    }
+}
+
+static void audio_print_settings (struct audsettings *as)
+{
+    dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
+
+    switch (as->fmt) {
+    case AUD_FMT_S8:
+        AUD_log (NULL, "S8");
+        break;
+    case AUD_FMT_U8:
+        AUD_log (NULL, "U8");
+        break;
+    case AUD_FMT_S16:
+        AUD_log (NULL, "S16");
+        break;
+    case AUD_FMT_U16:
+        AUD_log (NULL, "U16");
+        break;
+    case AUD_FMT_S32:
+        AUD_log (NULL, "S32");
+        break;
+    case AUD_FMT_U32:
+        AUD_log (NULL, "U32");
+        break;
+    default:
+        AUD_log (NULL, "invalid(%d)", as->fmt);
+        break;
+    }
+
+    AUD_log (NULL, " endianness=");
+    switch (as->endianness) {
+    case 0:
+        AUD_log (NULL, "little");
+        break;
+    case 1:
+        AUD_log (NULL, "big");
+        break;
+    default:
+        AUD_log (NULL, "invalid");
+        break;
+    }
+    AUD_log (NULL, "\n");
+}
+
+static int audio_validate_settings (struct audsettings *as)
+{
+    int invalid;
+
+    invalid = as->nchannels != 1 && as->nchannels != 2;
+    invalid |= as->endianness != 0 && as->endianness != 1;
+
+    switch (as->fmt) {
+    case AUD_FMT_S8:
+    case AUD_FMT_U8:
+    case AUD_FMT_S16:
+    case AUD_FMT_U16:
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        break;
+    default:
+        invalid = 1;
+        break;
+    }
+
+    invalid |= as->freq <= 0;
+    return invalid ? -1 : 0;
+}
+
+static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
+{
+    int bits = 8, sign = 0;
+
+    switch (as->fmt) {
+    case AUD_FMT_S8:
+        sign = 1;
+        /* fall through */
+    case AUD_FMT_U8:
+        break;
+
+    case AUD_FMT_S16:
+        sign = 1;
+        /* fall through */
+    case AUD_FMT_U16:
+        bits = 16;
+        break;
+
+    case AUD_FMT_S32:
+        sign = 1;
+        /* fall through */
+    case AUD_FMT_U32:
+        bits = 32;
+        break;
+    }
+    return info->freq == as->freq
+        && info->nchannels == as->nchannels
+        && info->sign == sign
+        && info->bits == bits
+        && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
+}
+
+void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
+{
+    int bits = 8, sign = 0, shift = 0;
+
+    switch (as->fmt) {
+    case AUD_FMT_S8:
+        sign = 1;
+    case AUD_FMT_U8:
+        break;
+
+    case AUD_FMT_S16:
+        sign = 1;
+    case AUD_FMT_U16:
+        bits = 16;
+        shift = 1;
+        break;
+
+    case AUD_FMT_S32:
+        sign = 1;
+    case AUD_FMT_U32:
+        bits = 32;
+        shift = 2;
+        break;
+    }
+
+    info->freq = as->freq;
+    info->bits = bits;
+    info->sign = sign;
+    info->nchannels = as->nchannels;
+    info->shift = (as->nchannels == 2) + shift;
+    info->align = (1 << info->shift) - 1;
+    info->bytes_per_second = info->freq << info->shift;
+    info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
+}
+
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
+{
+    if (!len) {
+        return;
+    }
+
+    if (info->sign) {
+        memset (buf, 0x00, len << info->shift);
+    }
+    else {
+        switch (info->bits) {
+        case 8:
+            memset (buf, 0x80, len << info->shift);
+            break;
+
+        case 16:
+            {
+                int i;
+                uint16_t *p = buf;
+                int shift = info->nchannels - 1;
+                short s = INT16_MAX;
+
+                if (info->swap_endianness) {
+                    s = bswap16 (s);
+                }
+
+                for (i = 0; i < len << shift; i++) {
+                    p[i] = s;
+                }
+            }
+            break;
+
+        case 32:
+            {
+                int i;
+                uint32_t *p = buf;
+                int shift = info->nchannels - 1;
+                int32_t s = INT32_MAX;
+
+                if (info->swap_endianness) {
+                    s = bswap32 (s);
+                }
+
+                for (i = 0; i < len << shift; i++) {
+                    p[i] = s;
+                }
+            }
+            break;
+
+        default:
+            AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
+                     info->bits);
+            break;
+        }
+    }
+}
+
+/*
+ * Capture
+ */
+static void noop_conv (struct st_sample *dst, const void *src, int samples)
+{
+    (void) src;
+    (void) dst;
+    (void) samples;
+}
+
+static CaptureVoiceOut *audio_pcm_capture_find_specific (
+    struct audsettings *as
+    )
+{
+    CaptureVoiceOut *cap;
+    AudioState *s = &glob_audio_state;
+
+    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+        if (audio_pcm_info_eq (&cap->hw.info, as)) {
+            return cap;
+        }
+    }
+    return NULL;
+}
+
+static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
+{
+    struct capture_callback *cb;
+
+#ifdef DEBUG_CAPTURE
+    dolog ("notification %d sent\n", cmd);
+#endif
+    for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+        cb->ops.notify (cb->opaque, cmd);
+    }
+}
+
+static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
+{
+    if (cap->hw.enabled != enabled) {
+        audcnotification_e cmd;
+        cap->hw.enabled = enabled;
+        cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
+        audio_notify_capture (cap, cmd);
+    }
+}
+
+static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
+{
+    HWVoiceOut *hw = &cap->hw;
+    SWVoiceOut *sw;
+    int enabled = 0;
+
+    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        if (sw->active) {
+            enabled = 1;
+            break;
+        }
+    }
+    audio_capture_maybe_changed (cap, enabled);
+}
+
+static void audio_detach_capture (HWVoiceOut *hw)
+{
+    SWVoiceCap *sc = hw->cap_head.lh_first;
+
+    while (sc) {
+        SWVoiceCap *sc1 = sc->entries.le_next;
+        SWVoiceOut *sw = &sc->sw;
+        CaptureVoiceOut *cap = sc->cap;
+        int was_active = sw->active;
+
+        if (sw->rate) {
+            st_rate_stop (sw->rate);
+            sw->rate = NULL;
+        }
+
+        QLIST_REMOVE (sw, entries);
+        QLIST_REMOVE (sc, entries);
+        g_free (sc);
+        if (was_active) {
+            /* We have removed soft voice from the capture:
+               this might have changed the overall status of the capture
+               since this might have been the only active voice */
+            audio_recalc_and_notify_capture (cap);
+        }
+        sc = sc1;
+    }
+}
+
+static int audio_attach_capture (HWVoiceOut *hw)
+{
+    AudioState *s = &glob_audio_state;
+    CaptureVoiceOut *cap;
+
+    audio_detach_capture (hw);
+    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+        SWVoiceCap *sc;
+        SWVoiceOut *sw;
+        HWVoiceOut *hw_cap = &cap->hw;
+
+        sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
+        if (!sc) {
+            dolog ("Could not allocate soft capture voice (%zu bytes)\n",
+                   sizeof (*sc));
+            return -1;
+        }
+
+        sc->cap = cap;
+        sw = &sc->sw;
+        sw->hw = hw_cap;
+        sw->info = hw->info;
+        sw->empty = 1;
+        sw->active = hw->enabled;
+        sw->conv = noop_conv;
+        sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
+        sw->vol = nominal_volume;
+        sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
+        if (!sw->rate) {
+            dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
+            g_free (sw);
+            return -1;
+        }
+        QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
+        QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
+#ifdef DEBUG_CAPTURE
+        sw->name = g_strdup_printf ("for %p %d,%d,%d",
+                                    hw, sw->info.freq, sw->info.bits,
+                                    sw->info.nchannels);
+        dolog ("Added %s active = %d\n", sw->name, sw->active);
+#endif
+        if (sw->active) {
+            audio_capture_maybe_changed (cap, 1);
+        }
+    }
+    return 0;
+}
+
+/*
+ * Hard voice (capture)
+ */
+static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
+{
+    SWVoiceIn *sw;
+    int m = hw->total_samples_captured;
+
+    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        if (sw->active) {
+            m = audio_MIN (m, sw->total_hw_samples_acquired);
+        }
+    }
+    return m;
+}
+
+int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
+{
+    int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+        return 0;
+    }
+    return live;
+}
+
+int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
+                           int live, int pending)
+{
+    int left = hw->samples - pending;
+    int len = audio_MIN (left, live);
+    int clipped = 0;
+
+    while (len) {
+        struct st_sample *src = hw->mix_buf + hw->rpos;
+        uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
+        int samples_till_end_of_buf = hw->samples - hw->rpos;
+        int samples_to_clip = audio_MIN (len, samples_till_end_of_buf);
+
+        hw->clip (dst, src, samples_to_clip);
+
+        hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
+        len -= samples_to_clip;
+        clipped += samples_to_clip;
+    }
+    return clipped;
+}
+
+/*
+ * Soft voice (capture)
+ */
+static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
+{
+    HWVoiceIn *hw = sw->hw;
+    int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+    int rpos;
+
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+        return 0;
+    }
+
+    rpos = hw->wpos - live;
+    if (rpos >= 0) {
+        return rpos;
+    }
+    else {
+        return hw->samples + rpos;
+    }
+}
+
+int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
+{
+    HWVoiceIn *hw = sw->hw;
+    int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+    struct st_sample *src, *dst = sw->buf;
+
+    rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
+
+    live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+        dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+        return 0;
+    }
+
+    samples = size >> sw->info.shift;
+    if (!live) {
+        return 0;
+    }
+
+    swlim = (live * sw->ratio) >> 32;
+    swlim = audio_MIN (swlim, samples);
+
+    while (swlim) {
+        src = hw->conv_buf + rpos;
+        isamp = hw->wpos - rpos;
+        /* XXX: <= ? */
+        if (isamp <= 0) {
+            isamp = hw->samples - rpos;
+        }
+
+        if (!isamp) {
+            break;
+        }
+        osamp = swlim;
+
+        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
+            dolog ("osamp=%d\n", osamp);
+            return 0;
+        }
+
+        st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
+        swlim -= osamp;
+        rpos = (rpos + isamp) % hw->samples;
+        dst += osamp;
+        ret += osamp;
+        total += isamp;
+    }
+
+    if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
+        mixeng_volume (sw->buf, ret, &sw->vol);
+    }
+
+    sw->clip (buf, sw->buf, ret);
+    sw->total_hw_samples_acquired += total;
+    return ret << sw->info.shift;
+}
+
+/*
+ * Hard voice (playback)
+ */
+static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
+{
+    SWVoiceOut *sw;
+    int m = INT_MAX;
+    int nb_live = 0;
+
+    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        if (sw->active || !sw->empty) {
+            m = audio_MIN (m, sw->total_hw_samples_mixed);
+            nb_live += 1;
+        }
+    }
+
+    *nb_livep = nb_live;
+    return m;
+}
+
+static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
+{
+    int smin;
+    int nb_live1;
+
+    smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
+    if (nb_live) {
+        *nb_live = nb_live1;
+    }
+
+    if (nb_live1) {
+        int live = smin;
+
+        if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+            dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+            return 0;
+        }
+        return live;
+    }
+    return 0;
+}
+
+/*
+ * Soft voice (playback)
+ */
+int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
+{
+    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+    int ret = 0, pos = 0, total = 0;
+
+    if (!sw) {
+        return size;
+    }
+
+    hwsamples = sw->hw->samples;
+
+    live = sw->total_hw_samples_mixed;
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
+        dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+        return 0;
+    }
+
+    if (live == hwsamples) {
+#ifdef DEBUG_OUT
+        dolog ("%s is full %d\n", sw->name, live);
+#endif
+        return 0;
+    }
+
+    wpos = (sw->hw->rpos + live) % hwsamples;
+    samples = size >> sw->info.shift;
+
+    dead = hwsamples - live;
+    swlim = ((int64_t) dead << 32) / sw->ratio;
+    swlim = audio_MIN (swlim, samples);
+    if (swlim) {
+        sw->conv (sw->buf, buf, swlim);
+
+        if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
+            mixeng_volume (sw->buf, swlim, &sw->vol);
+        }
+    }
+
+    while (swlim) {
+        dead = hwsamples - live;
+        left = hwsamples - wpos;
+        blck = audio_MIN (dead, left);
+        if (!blck) {
+            break;
+        }
+        isamp = swlim;
+        osamp = blck;
+        st_rate_flow_mix (
+            sw->rate,
+            sw->buf + pos,
+            sw->hw->mix_buf + wpos,
+            &isamp,
+            &osamp
+            );
+        ret += isamp;
+        swlim -= isamp;
+        pos += isamp;
+        live += osamp;
+        wpos = (wpos + osamp) % hwsamples;
+        total += osamp;
+    }
+
+    sw->total_hw_samples_mixed += total;
+    sw->empty = sw->total_hw_samples_mixed == 0;
+
+#ifdef DEBUG_OUT
+    dolog (
+        "%s: write size %d ret %d total sw %d\n",
+        SW_NAME (sw),
+        size >> sw->info.shift,
+        ret,
+        sw->total_hw_samples_mixed
+        );
+#endif
+
+    return ret << sw->info.shift;
+}
+
+#ifdef DEBUG_AUDIO
+static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
+{
+    dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
+           cap, info->bits, info->sign, info->freq, info->nchannels);
+}
+#endif
+
+#define DAC
+#include "audio_template.h"
+#undef DAC
+#include "audio_template.h"
+
+/*
+ * Timer
+ */
+static int audio_is_timer_needed (void)
+{
+    HWVoiceIn *hwi = NULL;
+    HWVoiceOut *hwo = NULL;
+
+    while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+        if (!hwo->poll_mode) return 1;
+    }
+    while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+        if (!hwi->poll_mode) return 1;
+    }
+    return 0;
+}
+
+static void audio_reset_timer (AudioState *s)
+{
+    if (audio_is_timer_needed ()) {
+        timer_mod (s->ts,
+            qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+    }
+    else {
+        timer_del (s->ts);
+    }
+}
+
+static void audio_timer (void *opaque)
+{
+    audio_run ("timer");
+    audio_reset_timer (opaque);
+}
+
+/*
+ * Public API
+ */
+int AUD_write (SWVoiceOut *sw, void *buf, int size)
+{
+    int bytes;
+
+    if (!sw) {
+        /* XXX: Consider options */
+        return size;
+    }
+
+    if (!sw->hw->enabled) {
+        dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
+        return 0;
+    }
+
+    bytes = sw->hw->pcm_ops->write (sw, buf, size);
+    return bytes;
+}
+
+int AUD_read (SWVoiceIn *sw, void *buf, int size)
+{
+    int bytes;
+
+    if (!sw) {
+        /* XXX: Consider options */
+        return size;
+    }
+
+    if (!sw->hw->enabled) {
+        dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
+        return 0;
+    }
+
+    bytes = sw->hw->pcm_ops->read (sw, buf, size);
+    return bytes;
+}
+
+int AUD_get_buffer_size_out (SWVoiceOut *sw)
+{
+    return sw->hw->samples << sw->hw->info.shift;
+}
+
+void AUD_set_active_out (SWVoiceOut *sw, int on)
+{
+    HWVoiceOut *hw;
+
+    if (!sw) {
+        return;
+    }
+
+    hw = sw->hw;
+    if (sw->active != on) {
+        AudioState *s = &glob_audio_state;
+        SWVoiceOut *temp_sw;
+        SWVoiceCap *sc;
+
+        if (on) {
+            hw->pending_disable = 0;
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                if (s->vm_running) {
+                    hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out);
+                    audio_reset_timer (s);
+                }
+            }
+        }
+        else {
+            if (hw->enabled) {
+                int nb_active = 0;
+
+                for (temp_sw = hw->sw_head.lh_first; temp_sw;
+                     temp_sw = temp_sw->entries.le_next) {
+                    nb_active += temp_sw->active != 0;
+                }
+
+                hw->pending_disable = nb_active == 1;
+            }
+        }
+
+        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+            sc->sw.active = hw->enabled;
+            if (hw->enabled) {
+                audio_capture_maybe_changed (sc->cap, 1);
+            }
+        }
+        sw->active = on;
+    }
+}
+
+void AUD_set_active_in (SWVoiceIn *sw, int on)
+{
+    HWVoiceIn *hw;
+
+    if (!sw) {
+        return;
+    }
+
+    hw = sw->hw;
+    if (sw->active != on) {
+        AudioState *s = &glob_audio_state;
+        SWVoiceIn *temp_sw;
+
+        if (on) {
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                if (s->vm_running) {
+                    hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in);
+                    audio_reset_timer (s);
+                }
+            }
+            sw->total_hw_samples_acquired = hw->total_samples_captured;
+        }
+        else {
+            if (hw->enabled) {
+                int nb_active = 0;
+
+                for (temp_sw = hw->sw_head.lh_first; temp_sw;
+                     temp_sw = temp_sw->entries.le_next) {
+                    nb_active += temp_sw->active != 0;
+                }
+
+                if (nb_active == 1) {
+                    hw->enabled = 0;
+                    hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
+                }
+            }
+        }
+        sw->active = on;
+    }
+}
+
+static int audio_get_avail (SWVoiceIn *sw)
+{
+    int live;
+
+    if (!sw) {
+        return 0;
+    }
+
+    live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
+        dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+        return 0;
+    }
+
+    ldebug (
+        "%s: get_avail live %d ret %" PRId64 "\n",
+        SW_NAME (sw),
+        live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+        );
+
+    return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+}
+
+static int audio_get_free (SWVoiceOut *sw)
+{
+    int live, dead;
+
+    if (!sw) {
+        return 0;
+    }
+
+    live = sw->total_hw_samples_mixed;
+
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
+        dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+        return 0;
+    }
+
+    dead = sw->hw->samples - live;
+
+#ifdef DEBUG_OUT
+    dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
+           SW_NAME (sw),
+           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+#endif
+
+    return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
+}
+
+static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
+{
+    int n;
+
+    if (hw->enabled) {
+        SWVoiceCap *sc;
+
+        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+            SWVoiceOut *sw = &sc->sw;
+            int rpos2 = rpos;
+
+            n = samples;
+            while (n) {
+                int till_end_of_hw = hw->samples - rpos2;
+                int to_write = audio_MIN (till_end_of_hw, n);
+                int bytes = to_write << hw->info.shift;
+                int written;
+
+                sw->buf = hw->mix_buf + rpos2;
+                written = audio_pcm_sw_write (sw, NULL, bytes);
+                if (written - bytes) {
+                    dolog ("Could not mix %d bytes into a capture "
+                           "buffer, mixed %d\n",
+                           bytes, written);
+                    break;
+                }
+                n -= to_write;
+                rpos2 = (rpos2 + to_write) % hw->samples;
+            }
+        }
+    }
+
+    n = audio_MIN (samples, hw->samples - rpos);
+    mixeng_clear (hw->mix_buf + rpos, n);
+    mixeng_clear (hw->mix_buf, samples - n);
+}
+
+static void audio_run_out (AudioState *s)
+{
+    HWVoiceOut *hw = NULL;
+    SWVoiceOut *sw;
+
+    while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
+        int played;
+        int live, free, nb_live, cleanup_required, prev_rpos;
+
+        live = audio_pcm_hw_get_live_out (hw, &nb_live);
+        if (!nb_live) {
+            live = 0;
+        }
+
+        if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+            dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+            continue;
+        }
+
+        if (hw->pending_disable && !nb_live) {
+            SWVoiceCap *sc;
+#ifdef DEBUG_OUT
+            dolog ("Disabling voice\n");
+#endif
+            hw->enabled = 0;
+            hw->pending_disable = 0;
+            hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+            for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+                sc->sw.active = 0;
+                audio_recalc_and_notify_capture (sc->cap);
+            }
+            continue;
+        }
+
+        if (!live) {
+            for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+                if (sw->active) {
+                    free = audio_get_free (sw);
+                    if (free > 0) {
+                        sw->callback.fn (sw->callback.opaque, free);
+                    }
+                }
+            }
+            continue;
+        }
+
+        prev_rpos = hw->rpos;
+        played = hw->pcm_ops->run_out (hw, live);
+        if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
+            dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
+                   hw->rpos, hw->samples, played);
+            hw->rpos = 0;
+        }
+
+#ifdef DEBUG_OUT
+        dolog ("played=%d\n", played);
+#endif
+
+        if (played) {
+            hw->ts_helper += played;
+            audio_capture_mix_and_clear (hw, prev_rpos, played);
+        }
+
+        cleanup_required = 0;
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            if (!sw->active && sw->empty) {
+                continue;
+            }
+
+            if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
+                dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
+                       played, sw->total_hw_samples_mixed);
+                played = sw->total_hw_samples_mixed;
+            }
+
+            sw->total_hw_samples_mixed -= played;
+
+            if (!sw->total_hw_samples_mixed) {
+                sw->empty = 1;
+                cleanup_required |= !sw->active && !sw->callback.fn;
+            }
+
+            if (sw->active) {
+                free = audio_get_free (sw);
+                if (free > 0) {
+                    sw->callback.fn (sw->callback.opaque, free);
+                }
+            }
+        }
+
+        if (cleanup_required) {
+            SWVoiceOut *sw1;
+
+            sw = hw->sw_head.lh_first;
+            while (sw) {
+                sw1 = sw->entries.le_next;
+                if (!sw->active && !sw->callback.fn) {
+                    audio_close_out (sw);
+                }
+                sw = sw1;
+            }
+        }
+    }
+}
+
+static void audio_run_in (AudioState *s)
+{
+    HWVoiceIn *hw = NULL;
+
+    while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
+        SWVoiceIn *sw;
+        int captured, min;
+
+        captured = hw->pcm_ops->run_in (hw);
+
+        min = audio_pcm_hw_find_min_in (hw);
+        hw->total_samples_captured += captured - min;
+        hw->ts_helper += captured;
+
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            sw->total_hw_samples_acquired -= min;
+
+            if (sw->active) {
+                int avail;
+
+                avail = audio_get_avail (sw);
+                if (avail > 0) {
+                    sw->callback.fn (sw->callback.opaque, avail);
+                }
+            }
+        }
+    }
+}
+
+static void audio_run_capture (AudioState *s)
+{
+    CaptureVoiceOut *cap;
+
+    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+        int live, rpos, captured;
+        HWVoiceOut *hw = &cap->hw;
+        SWVoiceOut *sw;
+
+        captured = live = audio_pcm_hw_get_live_out (hw, NULL);
+        rpos = hw->rpos;
+        while (live) {
+            int left = hw->samples - rpos;
+            int to_capture = audio_MIN (live, left);
+            struct st_sample *src;
+            struct capture_callback *cb;
+
+            src = hw->mix_buf + rpos;
+            hw->clip (cap->buf, src, to_capture);
+            mixeng_clear (src, to_capture);
+
+            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+                cb->ops.capture (cb->opaque, cap->buf,
+                                 to_capture << hw->info.shift);
+            }
+            rpos = (rpos + to_capture) % hw->samples;
+            live -= to_capture;
+        }
+        hw->rpos = rpos;
+
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            if (!sw->active && sw->empty) {
+                continue;
+            }
+
+            if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
+                dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
+                       captured, sw->total_hw_samples_mixed);
+                captured = sw->total_hw_samples_mixed;
+            }
+
+            sw->total_hw_samples_mixed -= captured;
+            sw->empty = sw->total_hw_samples_mixed == 0;
+        }
+    }
+}
+
+void audio_run (const char *msg)
+{
+    AudioState *s = &glob_audio_state;
+
+    audio_run_out (s);
+    audio_run_in (s);
+    audio_run_capture (s);
+#ifdef DEBUG_POLL
+    {
+        static double prevtime;
+        double currtime;
+        struct timeval tv;
+
+        if (gettimeofday (&tv, NULL)) {
+            perror ("audio_run: gettimeofday");
+            return;
+        }
+
+        currtime = tv.tv_sec + tv.tv_usec * 1e-6;
+        dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
+        prevtime = currtime;
+    }
+#endif
+}
+
+static struct audio_option audio_options[] = {
+    /* DAC */
+    {
+        .name  = "DAC_FIXED_SETTINGS",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &conf.fixed_out.enabled,
+        .descr = "Use fixed settings for host DAC"
+    },
+    {
+        .name  = "DAC_FIXED_FREQ",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.fixed_out.settings.freq,
+        .descr = "Frequency for fixed host DAC"
+    },
+    {
+        .name  = "DAC_FIXED_FMT",
+        .tag   = AUD_OPT_FMT,
+        .valp  = &conf.fixed_out.settings.fmt,
+        .descr = "Format for fixed host DAC"
+    },
+    {
+        .name  = "DAC_FIXED_CHANNELS",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.fixed_out.settings.nchannels,
+        .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
+    },
+    {
+        .name  = "DAC_VOICES",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.fixed_out.nb_voices,
+        .descr = "Number of voices for DAC"
+    },
+    {
+        .name  = "DAC_TRY_POLL",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &conf.try_poll_out,
+        .descr = "Attempt using poll mode for DAC"
+    },
+    /* ADC */
+    {
+        .name  = "ADC_FIXED_SETTINGS",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &conf.fixed_in.enabled,
+        .descr = "Use fixed settings for host ADC"
+    },
+    {
+        .name  = "ADC_FIXED_FREQ",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.fixed_in.settings.freq,
+        .descr = "Frequency for fixed host ADC"
+    },
+    {
+        .name  = "ADC_FIXED_FMT",
+        .tag   = AUD_OPT_FMT,
+        .valp  = &conf.fixed_in.settings.fmt,
+        .descr = "Format for fixed host ADC"
+    },
+    {
+        .name  = "ADC_FIXED_CHANNELS",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.fixed_in.settings.nchannels,
+        .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
+    },
+    {
+        .name  = "ADC_VOICES",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.fixed_in.nb_voices,
+        .descr = "Number of voices for ADC"
+    },
+    {
+        .name  = "ADC_TRY_POLL",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &conf.try_poll_in,
+        .descr = "Attempt using poll mode for ADC"
+    },
+    /* Misc */
+    {
+        .name  = "TIMER_PERIOD",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.period.hertz,
+        .descr = "Timer period in HZ (0 - use lowest possible)"
+    },
+    { /* End of list */ }
+};
+
+static void audio_pp_nb_voices (const char *typ, int nb)
+{
+    switch (nb) {
+    case 0:
+        printf ("Does not support %s\n", typ);
+        break;
+    case 1:
+        printf ("One %s voice\n", typ);
+        break;
+    case INT_MAX:
+        printf ("Theoretically supports many %s voices\n", typ);
+        break;
+    default:
+        printf ("Theoretically supports up to %d %s voices\n", nb, typ);
+        break;
+    }
+
+}
+
+void AUD_help (void)
+{
+    size_t i;
+
+    audio_process_options ("AUDIO", audio_options);
+    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+        struct audio_driver *d = drvtab[i];
+        if (d->options) {
+            audio_process_options (d->name, d->options);
+        }
+    }
+
+    printf ("Audio options:\n");
+    audio_print_options ("AUDIO", audio_options);
+    printf ("\n");
+
+    printf ("Available drivers:\n");
+
+    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+        struct audio_driver *d = drvtab[i];
+
+        printf ("Name: %s\n", d->name);
+        printf ("Description: %s\n", d->descr);
+
+        audio_pp_nb_voices ("playback", d->max_voices_out);
+        audio_pp_nb_voices ("capture", d->max_voices_in);
+
+        if (d->options) {
+            printf ("Options:\n");
+            audio_print_options (d->name, d->options);
+        }
+        else {
+            printf ("No options\n");
+        }
+        printf ("\n");
+    }
+
+    printf (
+        "Options are settable through environment variables.\n"
+        "Example:\n"
+#ifdef _WIN32
+        "  set QEMU_AUDIO_DRV=wav\n"
+        "  set QEMU_WAV_PATH=c:\\tune.wav\n"
+#else
+        "  export QEMU_AUDIO_DRV=wav\n"
+        "  export QEMU_WAV_PATH=$HOME/tune.wav\n"
+        "(for csh replace export with setenv in the above)\n"
+#endif
+        "  qemu ...\n\n"
+        );
+}
+
+static int audio_driver_init (AudioState *s, struct audio_driver *drv)
+{
+    if (drv->options) {
+        audio_process_options (drv->name, drv->options);
+    }
+    s->drv_opaque = drv->init ();
+
+    if (s->drv_opaque) {
+        audio_init_nb_voices_out (drv);
+        audio_init_nb_voices_in (drv);
+        s->drv = drv;
+        return 0;
+    }
+    else {
+        dolog ("Could not init `%s' audio driver\n", drv->name);
+        return -1;
+    }
+}
+
+static void audio_vm_change_state_handler (void *opaque, int running,
+                                           RunState state)
+{
+    AudioState *s = opaque;
+    HWVoiceOut *hwo = NULL;
+    HWVoiceIn *hwi = NULL;
+    int op = running ? VOICE_ENABLE : VOICE_DISABLE;
+
+    s->vm_running = running;
+    while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+        hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
+    }
+
+    while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+        hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
+    }
+    audio_reset_timer (s);
+}
+
+static void audio_atexit (void)
+{
+    AudioState *s = &glob_audio_state;
+    HWVoiceOut *hwo = NULL;
+    HWVoiceIn *hwi = NULL;
+
+    while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+        SWVoiceCap *sc;
+
+        if (hwo->enabled) {
+            hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
+        }
+        hwo->pcm_ops->fini_out (hwo);
+
+        for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+            CaptureVoiceOut *cap = sc->cap;
+            struct capture_callback *cb;
+
+            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+                cb->ops.destroy (cb->opaque);
+            }
+        }
+    }
+
+    while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+        if (hwi->enabled) {
+            hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
+        }
+        hwi->pcm_ops->fini_in (hwi);
+    }
+
+    if (s->drv) {
+        s->drv->fini (s->drv_opaque);
+    }
+}
+
+static const VMStateDescription vmstate_audio = {
+    .name = "audio",
+    .version_id = 1,
+    .minimum_version_id = 1,
+    .fields = (VMStateField[]) {
+        VMSTATE_END_OF_LIST()
+    }
+};
+
+static void audio_init (void)
+{
+    size_t i;
+    int done = 0;
+    const char *drvname;
+    VMChangeStateEntry *e;
+    AudioState *s = &glob_audio_state;
+
+    if (s->drv) {
+        return;
+    }
+
+    QLIST_INIT (&s->hw_head_out);
+    QLIST_INIT (&s->hw_head_in);
+    QLIST_INIT (&s->cap_head);
+    atexit (audio_atexit);
+
+    s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
+    if (!s->ts) {
+        hw_error("Could not create audio timer\n");
+    }
+
+    audio_process_options ("AUDIO", audio_options);
+
+    s->nb_hw_voices_out = conf.fixed_out.nb_voices;
+    s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+
+    if (s->nb_hw_voices_out <= 0) {
+        dolog ("Bogus number of playback voices %d, setting to 1\n",
+               s->nb_hw_voices_out);
+        s->nb_hw_voices_out = 1;
+    }
+
+    if (s->nb_hw_voices_in <= 0) {
+        dolog ("Bogus number of capture voices %d, setting to 0\n",
+               s->nb_hw_voices_in);
+        s->nb_hw_voices_in = 0;
+    }
+
+    {
+        int def;
+        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
+    }
+
+    if (drvname) {
+        int found = 0;
+
+        for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+            if (!strcmp (drvname, drvtab[i]->name)) {
+                done = !audio_driver_init (s, drvtab[i]);
+                found = 1;
+                break;
+            }
+        }
+
+        if (!found) {
+            dolog ("Unknown audio driver `%s'\n", drvname);
+            dolog ("Run with -audio-help to list available drivers\n");
+        }
+    }
+
+    if (!done) {
+        for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
+            if (drvtab[i]->can_be_default) {
+                done = !audio_driver_init (s, drvtab[i]);
+            }
+        }
+    }
+
+    if (!done) {
+        done = !audio_driver_init (s, &no_audio_driver);
+        if (!done) {
+            hw_error("Could not initialize audio subsystem\n");
+        }
+        else {
+            dolog ("warning: Using timer based audio emulation\n");
+        }
+    }
+
+    if (conf.period.hertz <= 0) {
+        if (conf.period.hertz < 0) {
+            dolog ("warning: Timer period is negative - %d "
+                   "treating as zero\n",
+                   conf.period.hertz);
+        }
+        conf.period.ticks = 1;
+    } else {
+        conf.period.ticks =
+            muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
+    }
+
+    e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
+    if (!e) {
+        dolog ("warning: Could not register change state handler\n"
+               "(Audio can continue looping even after stopping the VM)\n");
+    }
+
+    QLIST_INIT (&s->card_head);
+    vmstate_register (NULL, 0, &vmstate_audio, s);
+}
+
+void AUD_register_card (const char *name, QEMUSoundCard *card)
+{
+    audio_init ();
+    card->name = g_strdup (name);
+    memset (&card->entries, 0, sizeof (card->entries));
+    QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
+}
+
+void AUD_remove_card (QEMUSoundCard *card)
+{
+    QLIST_REMOVE (card, entries);
+    g_free (card->name);
+}
+
+
+CaptureVoiceOut *AUD_add_capture (
+    struct audsettings *as,
+    struct audio_capture_ops *ops,
+    void *cb_opaque
+    )
+{
+    AudioState *s = &glob_audio_state;
+    CaptureVoiceOut *cap;
+    struct capture_callback *cb;
+
+    if (audio_validate_settings (as)) {
+        dolog ("Invalid settings were passed when trying to add capture\n");
+        audio_print_settings (as);
+        goto err0;
+    }
+
+    cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
+    if (!cb) {
+        dolog ("Could not allocate capture callback information, size %zu\n",
+               sizeof (*cb));
+        goto err0;
+    }
+    cb->ops = *ops;
+    cb->opaque = cb_opaque;
+
+    cap = audio_pcm_capture_find_specific (as);
+    if (cap) {
+        QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+        return cap;
+    }
+    else {
+        HWVoiceOut *hw;
+        CaptureVoiceOut *cap;
+
+        cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
+        if (!cap) {
+            dolog ("Could not allocate capture voice, size %zu\n",
+                   sizeof (*cap));
+            goto err1;
+        }
+
+        hw = &cap->hw;
+        QLIST_INIT (&hw->sw_head);
+        QLIST_INIT (&cap->cb_head);
+
+        /* XXX find a more elegant way */
+        hw->samples = 4096 * 4;
+        hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
+                                    sizeof (struct st_sample));
+        if (!hw->mix_buf) {
+            dolog ("Could not allocate capture mix buffer (%d samples)\n",
+                   hw->samples);
+            goto err2;
+        }
+
+        audio_pcm_init_info (&hw->info, as);
+
+        cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+        if (!cap->buf) {
+            dolog ("Could not allocate capture buffer "
+                   "(%d samples, each %d bytes)\n",
+                   hw->samples, 1 << hw->info.shift);
+            goto err3;
+        }
+
+        hw->clip = mixeng_clip
+            [hw->info.nchannels == 2]
+            [hw->info.sign]
+            [hw->info.swap_endianness]
+            [audio_bits_to_index (hw->info.bits)];
+
+        QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
+        QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+
+        hw = NULL;
+        while ((hw = audio_pcm_hw_find_any_out (hw))) {
+            audio_attach_capture (hw);
+        }
+        return cap;
+
+    err3:
+        g_free (cap->hw.mix_buf);
+    err2:
+        g_free (cap);
+    err1:
+        g_free (cb);
+    err0:
+        return NULL;
+    }
+}
+
+void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
+{
+    struct capture_callback *cb;
+
+    for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+        if (cb->opaque == cb_opaque) {
+            cb->ops.destroy (cb_opaque);
+            QLIST_REMOVE (cb, entries);
+            g_free (cb);
+
+            if (!cap->cb_head.lh_first) {
+                SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
+
+                while (sw) {
+                    SWVoiceCap *sc = (SWVoiceCap *) sw;
+#ifdef DEBUG_CAPTURE
+                    dolog ("freeing %s\n", sw->name);
+#endif
+
+                    sw1 = sw->entries.le_next;
+                    if (sw->rate) {
+                        st_rate_stop (sw->rate);
+                        sw->rate = NULL;
+                    }
+                    QLIST_REMOVE (sw, entries);
+                    QLIST_REMOVE (sc, entries);
+                    g_free (sc);
+                    sw = sw1;
+                }
+                QLIST_REMOVE (cap, entries);
+                g_free (cap);
+            }
+            return;
+        }
+    }
+}
+
+void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    if (sw) {
+        HWVoiceOut *hw = sw->hw;
+
+        sw->vol.mute = mute;
+        sw->vol.l = nominal_volume.l * lvol / 255;
+        sw->vol.r = nominal_volume.r * rvol / 255;
+
+        if (hw->pcm_ops->ctl_out) {
+            hw->pcm_ops->ctl_out (hw, VOICE_VOLUME, sw);
+        }
+    }
+}
+
+void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    if (sw) {
+        HWVoiceIn *hw = sw->hw;
+
+        sw->vol.mute = mute;
+        sw->vol.l = nominal_volume.l * lvol / 255;
+        sw->vol.r = nominal_volume.r * rvol / 255;
+
+        if (hw->pcm_ops->ctl_in) {
+            hw->pcm_ops->ctl_in (hw, VOICE_VOLUME, sw);
+        }
+    }
+}
diff --git a/qemu/audio/audio.h b/qemu/audio/audio.h
new file mode 100644
index 000000000..e7ea39777
--- /dev/null
+++ b/qemu/audio/audio.h
@@ -0,0 +1,166 @@
+/*
+ * QEMU Audio subsystem header
+ *
+ * Copyright (c) 2003-2005 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.
+ */
+#ifndef QEMU_AUDIO_H
+#define QEMU_AUDIO_H
+
+#include "config-host.h"
+#include "qemu/queue.h"
+
+typedef void (*audio_callback_fn) (void *opaque, int avail);
+
+typedef enum {
+    AUD_FMT_U8,
+    AUD_FMT_S8,
+    AUD_FMT_U16,
+    AUD_FMT_S16,
+    AUD_FMT_U32,
+    AUD_FMT_S32
+} audfmt_e;
+
+#ifdef HOST_WORDS_BIGENDIAN
+#define AUDIO_HOST_ENDIANNESS 1
+#else
+#define AUDIO_HOST_ENDIANNESS 0
+#endif
+
+struct audsettings {
+    int freq;
+    int nchannels;
+    audfmt_e fmt;
+    int endianness;
+};
+
+typedef enum {
+    AUD_CNOTIFY_ENABLE,
+    AUD_CNOTIFY_DISABLE
+} audcnotification_e;
+
+struct audio_capture_ops {
+    void (*notify) (void *opaque, audcnotification_e cmd);
+    void (*capture) (void *opaque, void *buf, int size);
+    void (*destroy) (void *opaque);
+};
+
+struct capture_ops {
+    void (*info) (void *opaque);
+    void (*destroy) (void *opaque);
+};
+
+typedef struct CaptureState {
+    void *opaque;
+    struct capture_ops ops;
+    QLIST_ENTRY (CaptureState) entries;
+} CaptureState;
+
+typedef struct SWVoiceOut SWVoiceOut;
+typedef struct CaptureVoiceOut CaptureVoiceOut;
+typedef struct SWVoiceIn SWVoiceIn;
+
+typedef struct QEMUSoundCard {
+    char *name;
+    QLIST_ENTRY (QEMUSoundCard) entries;
+} QEMUSoundCard;
+
+typedef struct QEMUAudioTimeStamp {
+    uint64_t old_ts;
+} QEMUAudioTimeStamp;
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap) GCC_FMT_ATTR(2, 0);
+void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
+
+void AUD_help (void);
+void AUD_register_card (const char *name, QEMUSoundCard *card);
+void AUD_remove_card (QEMUSoundCard *card);
+CaptureVoiceOut *AUD_add_capture (
+    struct audsettings *as,
+    struct audio_capture_ops *ops,
+    void *opaque
+    );
+void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque);
+
+SWVoiceOut *AUD_open_out (
+    QEMUSoundCard *card,
+    SWVoiceOut *sw,
+    const char *name,
+    void *callback_opaque,
+    audio_callback_fn callback_fn,
+    struct audsettings *settings
+    );
+
+void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
+int  AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+int  AUD_get_buffer_size_out (SWVoiceOut *sw);
+void AUD_set_active_out (SWVoiceOut *sw, int on);
+int  AUD_is_active_out (SWVoiceOut *sw);
+
+void     AUD_init_time_stamp_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
+uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
+
+void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol);
+void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol);
+
+SWVoiceIn *AUD_open_in (
+    QEMUSoundCard *card,
+    SWVoiceIn *sw,
+    const char *name,
+    void *callback_opaque,
+    audio_callback_fn callback_fn,
+    struct audsettings *settings
+    );
+
+void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
+int  AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+void AUD_set_active_in (SWVoiceIn *sw, int on);
+int  AUD_is_active_in (SWVoiceIn *sw);
+
+void     AUD_init_time_stamp_in (SWVoiceIn *sw, QEMUAudioTimeStamp *ts);
+uint64_t AUD_get_elapsed_usec_in (SWVoiceIn *sw, QEMUAudioTimeStamp *ts);
+
+static inline void *advance (void *p, int incr)
+{
+    uint8_t *d = p;
+    return (d + incr);
+}
+
+#ifdef __GNUC__
+#define audio_MIN(a, b) ( __extension__ ({      \
+    __typeof (a) ta = a;                        \
+    __typeof (b) tb = b;                        \
+    ((ta)>(tb)?(tb):(ta));                      \
+}))
+
+#define audio_MAX(a, b) ( __extension__ ({      \
+    __typeof (a) ta = a;                        \
+    __typeof (b) tb = b;                        \
+    ((ta)<(tb)?(tb):(ta));                      \
+}))
+#else
+#define audio_MIN(a, b) ((a)>(b)?(b):(a))
+#define audio_MAX(a, b) ((a)<(b)?(b):(a))
+#endif
+
+int wav_start_capture (CaptureState *s, const char *path, int freq,
+                       int bits, int nchannels);
+
+#endif  /* audio.h */
diff --git a/qemu/audio/audio_int.h b/qemu/audio/audio_int.h
new file mode 100644
index 000000000..566df5edf
--- /dev/null
+++ b/qemu/audio/audio_int.h
@@ -0,0 +1,260 @@
+/*
+ * QEMU Audio subsystem header
+ *
+ * Copyright (c) 2003-2005 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.
+ */
+#ifndef QEMU_AUDIO_INT_H
+#define QEMU_AUDIO_INT_H
+
+#ifdef CONFIG_COREAUDIO
+#define FLOAT_MIXENG
+/* #define RECIPROCAL */
+#endif
+#include "mixeng.h"
+
+struct audio_pcm_ops;
+
+typedef enum {
+    AUD_OPT_INT,
+    AUD_OPT_FMT,
+    AUD_OPT_STR,
+    AUD_OPT_BOOL
+} audio_option_tag_e;
+
+struct audio_option {
+    const char *name;
+    audio_option_tag_e tag;
+    void *valp;
+    const char *descr;
+    int *overriddenp;
+    int overridden;
+};
+
+struct audio_callback {
+    void *opaque;
+    audio_callback_fn fn;
+};
+
+struct audio_pcm_info {
+    int bits;
+    int sign;
+    int freq;
+    int nchannels;
+    int align;
+    int shift;
+    int bytes_per_second;
+    int swap_endianness;
+};
+
+typedef struct SWVoiceCap SWVoiceCap;
+
+typedef struct HWVoiceOut {
+    int enabled;
+    int poll_mode;
+    int pending_disable;
+    struct audio_pcm_info info;
+
+    f_sample *clip;
+
+    int rpos;
+    uint64_t ts_helper;
+
+    struct st_sample *mix_buf;
+
+    int samples;
+    QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
+    QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
+    int ctl_caps;
+    struct audio_pcm_ops *pcm_ops;
+    QLIST_ENTRY (HWVoiceOut) entries;
+} HWVoiceOut;
+
+typedef struct HWVoiceIn {
+    int enabled;
+    int poll_mode;
+    struct audio_pcm_info info;
+
+    t_sample *conv;
+
+    int wpos;
+    int total_samples_captured;
+    uint64_t ts_helper;
+
+    struct st_sample *conv_buf;
+
+    int samples;
+    QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
+    int ctl_caps;
+    struct audio_pcm_ops *pcm_ops;
+    QLIST_ENTRY (HWVoiceIn) entries;
+} HWVoiceIn;
+
+struct SWVoiceOut {
+    QEMUSoundCard *card;
+    struct audio_pcm_info info;
+    t_sample *conv;
+    int64_t ratio;
+    struct st_sample *buf;
+    void *rate;
+    int total_hw_samples_mixed;
+    int active;
+    int empty;
+    HWVoiceOut *hw;
+    char *name;
+    struct mixeng_volume vol;
+    struct audio_callback callback;
+    QLIST_ENTRY (SWVoiceOut) entries;
+};
+
+struct SWVoiceIn {
+    QEMUSoundCard *card;
+    int active;
+    struct audio_pcm_info info;
+    int64_t ratio;
+    void *rate;
+    int total_hw_samples_acquired;
+    struct st_sample *buf;
+    f_sample *clip;
+    HWVoiceIn *hw;
+    char *name;
+    struct mixeng_volume vol;
+    struct audio_callback callback;
+    QLIST_ENTRY (SWVoiceIn) entries;
+};
+
+struct audio_driver {
+    const char *name;
+    const char *descr;
+    struct audio_option *options;
+    void *(*init) (void);
+    void (*fini) (void *);
+    struct audio_pcm_ops *pcm_ops;
+    int can_be_default;
+    int max_voices_out;
+    int max_voices_in;
+    int voice_size_out;
+    int voice_size_in;
+    int ctl_caps;
+};
+
+struct audio_pcm_ops {
+    int  (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
+    void (*fini_out)(HWVoiceOut *hw);
+    int  (*run_out) (HWVoiceOut *hw, int live);
+    int  (*write)   (SWVoiceOut *sw, void *buf, int size);
+    int  (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+
+    int  (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
+    void (*fini_in) (HWVoiceIn *hw);
+    int  (*run_in)  (HWVoiceIn *hw);
+    int  (*read)    (SWVoiceIn *sw, void *buf, int size);
+    int  (*ctl_in)  (HWVoiceIn *hw, int cmd, ...);
+};
+
+struct capture_callback {
+    struct audio_capture_ops ops;
+    void *opaque;
+    QLIST_ENTRY (capture_callback) entries;
+};
+
+struct CaptureVoiceOut {
+    HWVoiceOut hw;
+    void *buf;
+    QLIST_HEAD (cb_listhead, capture_callback) cb_head;
+    QLIST_ENTRY (CaptureVoiceOut) entries;
+};
+
+struct SWVoiceCap {
+    SWVoiceOut sw;
+    CaptureVoiceOut *cap;
+    QLIST_ENTRY (SWVoiceCap) entries;
+};
+
+struct AudioState {
+    struct audio_driver *drv;
+    void *drv_opaque;
+
+    QEMUTimer *ts;
+    QLIST_HEAD (card_listhead, QEMUSoundCard) card_head;
+    QLIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
+    QLIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
+    QLIST_HEAD (cap_listhead, CaptureVoiceOut) cap_head;
+    int nb_hw_voices_out;
+    int nb_hw_voices_in;
+    int vm_running;
+};
+
+extern struct audio_driver no_audio_driver;
+extern struct audio_driver oss_audio_driver;
+extern struct audio_driver sdl_audio_driver;
+extern struct audio_driver wav_audio_driver;
+extern struct audio_driver alsa_audio_driver;
+extern struct audio_driver coreaudio_audio_driver;
+extern struct audio_driver dsound_audio_driver;
+extern struct audio_driver pa_audio_driver;
+extern struct audio_driver spice_audio_driver;
+extern const struct mixeng_volume nominal_volume;
+
+void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
+
+int  audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
+int  audio_pcm_hw_get_live_in (HWVoiceIn *hw);
+
+int  audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
+
+int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
+                           int live, int pending);
+
+int audio_bug (const char *funcname, int cond);
+void *audio_calloc (const char *funcname, int nmemb, size_t size);
+
+void audio_run (const char *msg);
+
+#define VOICE_ENABLE 1
+#define VOICE_DISABLE 2
+#define VOICE_VOLUME 3
+
+#define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
+
+static inline int audio_ring_dist (int dst, int src, int len)
+{
+    return (dst >= src) ? (dst - src) : (len - src + dst);
+}
+
+#define dolog(fmt, ...) AUD_log(AUDIO_CAP, fmt, ## __VA_ARGS__)
+
+#ifdef DEBUG
+#define ldebug(fmt, ...) AUD_log(AUDIO_CAP, fmt, ## __VA_ARGS__)
+#else
+#define ldebug(fmt, ...) (void)0
+#endif
+
+#define AUDIO_STRINGIFY_(n) #n
+#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
+
+#if defined _MSC_VER || defined __GNUC__
+#define AUDIO_FUNC __FUNCTION__
+#else
+#define AUDIO_FUNC __FILE__ ":" AUDIO_STRINGIFY (__LINE__)
+#endif
+
+#endif /* audio_int.h */
diff --git a/qemu/audio/audio_pt_int.c b/qemu/audio/audio_pt_int.c
new file mode 100644
index 000000000..9a9c306a9
--- /dev/null
+++ b/qemu/audio/audio_pt_int.c
@@ -0,0 +1,173 @@
+#include "qemu-common.h"
+#include "audio.h"
+
+#define AUDIO_CAP "audio-pt"
+
+#include "audio_int.h"
+#include "audio_pt_int.h"
+
+static void GCC_FMT_ATTR(3, 4) logerr (struct audio_pt *pt, int err,
+                                       const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (pt->drv, fmt, ap);
+    va_end (ap);
+
+    AUD_log (NULL, "\n");
+    AUD_log (pt->drv, "Reason: %s\n", strerror (err));
+}
+
+int audio_pt_init (struct audio_pt *p, void *(*func) (void *),
+                   void *opaque, const char *drv, const char *cap)
+{
+    int err, err2;
+    const char *efunc;
+    sigset_t set, old_set;
+
+    p->drv = drv;
+
+    err = sigfillset (&set);
+    if (err) {
+        logerr (p, errno, "%s(%s): sigfillset failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+
+    err = pthread_mutex_init (&p->mutex, NULL);
+    if (err) {
+        efunc = "pthread_mutex_init";
+        goto err0;
+    }
+
+    err = pthread_cond_init (&p->cond, NULL);
+    if (err) {
+        efunc = "pthread_cond_init";
+        goto err1;
+    }
+
+    err = pthread_sigmask (SIG_BLOCK, &set, &old_set);
+    if (err) {
+        efunc = "pthread_sigmask";
+        goto err2;
+    }
+
+    err = pthread_create (&p->thread, NULL, func, opaque);
+
+    err2 = pthread_sigmask (SIG_SETMASK, &old_set, NULL);
+    if (err2) {
+        logerr (p, err2, "%s(%s): pthread_sigmask (restore) failed",
+                cap, AUDIO_FUNC);
+        /* We have failed to restore original signal mask, all bets are off,
+           so terminate the process */
+        exit (EXIT_FAILURE);
+    }
+
+    if (err) {
+        efunc = "pthread_create";
+        goto err2;
+    }
+
+    return 0;
+
+ err2:
+    err2 = pthread_cond_destroy (&p->cond);
+    if (err2) {
+        logerr (p, err2, "%s(%s): pthread_cond_destroy failed", cap, AUDIO_FUNC);
+    }
+
+ err1:
+    err2 = pthread_mutex_destroy (&p->mutex);
+    if (err2) {
+        logerr (p, err2, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC);
+    }
+
+ err0:
+    logerr (p, err, "%s(%s): %s failed", cap, AUDIO_FUNC, efunc);
+    return -1;
+}
+
+int audio_pt_fini (struct audio_pt *p, const char *cap)
+{
+    int err, ret = 0;
+
+    err = pthread_cond_destroy (&p->cond);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_cond_destroy failed", cap, AUDIO_FUNC);
+        ret = -1;
+    }
+
+    err = pthread_mutex_destroy (&p->mutex);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC);
+        ret = -1;
+    }
+    return ret;
+}
+
+int audio_pt_lock (struct audio_pt *p, const char *cap)
+{
+    int err;
+
+    err = pthread_mutex_lock (&p->mutex);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_mutex_lock failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+    return 0;
+}
+
+int audio_pt_unlock (struct audio_pt *p, const char *cap)
+{
+    int err;
+
+    err = pthread_mutex_unlock (&p->mutex);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_mutex_unlock failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+    return 0;
+}
+
+int audio_pt_wait (struct audio_pt *p, const char *cap)
+{
+    int err;
+
+    err = pthread_cond_wait (&p->cond, &p->mutex);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_cond_wait failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+    return 0;
+}
+
+int audio_pt_unlock_and_signal (struct audio_pt *p, const char *cap)
+{
+    int err;
+
+    err = pthread_mutex_unlock (&p->mutex);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_mutex_unlock failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+    err = pthread_cond_signal (&p->cond);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_cond_signal failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+    return 0;
+}
+
+int audio_pt_join (struct audio_pt *p, void **arg, const char *cap)
+{
+    int err;
+    void *ret;
+
+    err = pthread_join (p->thread, &ret);
+    if (err) {
+        logerr (p, err, "%s(%s): pthread_join failed", cap, AUDIO_FUNC);
+        return -1;
+    }
+    *arg = ret;
+    return 0;
+}
diff --git a/qemu/audio/audio_pt_int.h b/qemu/audio/audio_pt_int.h
new file mode 100644
index 000000000..0dfff76aa
--- /dev/null
+++ b/qemu/audio/audio_pt_int.h
@@ -0,0 +1,22 @@
+#ifndef QEMU_AUDIO_PT_INT_H
+#define QEMU_AUDIO_PT_INT_H
+
+#include <pthread.h>
+
+struct audio_pt {
+    const char *drv;
+    pthread_t thread;
+    pthread_cond_t cond;
+    pthread_mutex_t mutex;
+};
+
+int audio_pt_init (struct audio_pt *, void *(*) (void *), void *,
+                   const char *, const char *);
+int audio_pt_fini (struct audio_pt *, const char *);
+int audio_pt_lock (struct audio_pt *, const char *);
+int audio_pt_unlock (struct audio_pt *, const char *);
+int audio_pt_wait (struct audio_pt *, const char *);
+int audio_pt_unlock_and_signal (struct audio_pt *, const char *);
+int audio_pt_join (struct audio_pt *, void **, const char *);
+
+#endif /* audio_pt_int.h */
diff --git a/qemu/audio/audio_template.h b/qemu/audio/audio_template.h
new file mode 100644
index 000000000..99b27b285
--- /dev/null
+++ b/qemu/audio/audio_template.h
@@ -0,0 +1,514 @@
+/*
+ * QEMU Audio subsystem header
+ *
+ * Copyright (c) 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.
+ */
+
+#ifdef DAC
+#define NAME "playback"
+#define HWBUF hw->mix_buf
+#define TYPE out
+#define HW HWVoiceOut
+#define SW SWVoiceOut
+#else
+#define NAME "capture"
+#define TYPE in
+#define HW HWVoiceIn
+#define SW SWVoiceIn
+#define HWBUF hw->conv_buf
+#endif
+
+static void glue (audio_init_nb_voices_, TYPE) (struct audio_driver *drv)
+{
+    AudioState *s = &glob_audio_state;
+    int max_voices = glue (drv->max_voices_, TYPE);
+    int voice_size = glue (drv->voice_size_, TYPE);
+
+    if (glue (s->nb_hw_voices_, TYPE) > max_voices) {
+        if (!max_voices) {
+#ifdef DAC
+            dolog ("Driver `%s' does not support " NAME "\n", drv->name);
+#endif
+        }
+        else {
+            dolog ("Driver `%s' does not support %d " NAME " voices, max %d\n",
+                   drv->name,
+                   glue (s->nb_hw_voices_, TYPE),
+                   max_voices);
+        }
+        glue (s->nb_hw_voices_, TYPE) = max_voices;
+    }
+
+    if (audio_bug (AUDIO_FUNC, !voice_size && max_voices)) {
+        dolog ("drv=`%s' voice_size=0 max_voices=%d\n",
+               drv->name, max_voices);
+        glue (s->nb_hw_voices_, TYPE) = 0;
+    }
+
+    if (audio_bug (AUDIO_FUNC, voice_size && !max_voices)) {
+        dolog ("drv=`%s' voice_size=%d max_voices=0\n",
+               drv->name, voice_size);
+    }
+}
+
+static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
+{
+    g_free (HWBUF);
+    HWBUF = NULL;
+}
+
+static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
+{
+    HWBUF = audio_calloc (AUDIO_FUNC, hw->samples, sizeof (struct st_sample));
+    if (!HWBUF) {
+        dolog ("Could not allocate " NAME " buffer (%d samples)\n",
+               hw->samples);
+        return -1;
+    }
+
+    return 0;
+}
+
+static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
+{
+    g_free (sw->buf);
+
+    if (sw->rate) {
+        st_rate_stop (sw->rate);
+    }
+
+    sw->buf = NULL;
+    sw->rate = NULL;
+}
+
+static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
+{
+    int samples;
+
+    samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
+
+    sw->buf = audio_calloc (AUDIO_FUNC, samples, sizeof (struct st_sample));
+    if (!sw->buf) {
+        dolog ("Could not allocate buffer for `%s' (%d samples)\n",
+               SW_NAME (sw), samples);
+        return -1;
+    }
+
+#ifdef DAC
+    sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
+#else
+    sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
+#endif
+    if (!sw->rate) {
+        g_free (sw->buf);
+        sw->buf = NULL;
+        return -1;
+    }
+    return 0;
+}
+
+static int glue (audio_pcm_sw_init_, TYPE) (
+    SW *sw,
+    HW *hw,
+    const char *name,
+    struct audsettings *as
+    )
+{
+    int err;
+
+    audio_pcm_init_info (&sw->info, as);
+    sw->hw = hw;
+    sw->active = 0;
+#ifdef DAC
+    sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
+    sw->total_hw_samples_mixed = 0;
+    sw->empty = 1;
+#else
+    sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
+#endif
+
+#ifdef DAC
+    sw->conv = mixeng_conv
+#else
+    sw->clip = mixeng_clip
+#endif
+        [sw->info.nchannels == 2]
+        [sw->info.sign]
+        [sw->info.swap_endianness]
+        [audio_bits_to_index (sw->info.bits)];
+
+    sw->name = g_strdup (name);
+    err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
+    if (err) {
+        g_free (sw->name);
+        sw->name = NULL;
+    }
+    return err;
+}
+
+static void glue (audio_pcm_sw_fini_, TYPE) (SW *sw)
+{
+    glue (audio_pcm_sw_free_resources_, TYPE) (sw);
+    g_free (sw->name);
+    sw->name = NULL;
+}
+
+static void glue (audio_pcm_hw_add_sw_, TYPE) (HW *hw, SW *sw)
+{
+    QLIST_INSERT_HEAD (&hw->sw_head, sw, entries);
+}
+
+static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
+{
+    QLIST_REMOVE (sw, entries);
+}
+
+static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
+{
+    AudioState *s = &glob_audio_state;
+    HW *hw = *hwp;
+
+    if (!hw->sw_head.lh_first) {
+#ifdef DAC
+        audio_detach_capture (hw);
+#endif
+        QLIST_REMOVE (hw, entries);
+        glue (hw->pcm_ops->fini_, TYPE) (hw);
+        glue (s->nb_hw_voices_, TYPE) += 1;
+        glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
+        g_free (hw);
+        *hwp = NULL;
+    }
+}
+
+static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+{
+    AudioState *s = &glob_audio_state;
+    return hw ? hw->entries.le_next : glue (s->hw_head_, TYPE).lh_first;
+}
+
+static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (hw->enabled) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
+    HW *hw,
+    struct audsettings *as
+    )
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (audio_pcm_info_eq (&hw->info, as)) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
+{
+    HW *hw;
+    AudioState *s = &glob_audio_state;
+    struct audio_driver *drv = s->drv;
+
+    if (!glue (s->nb_hw_voices_, TYPE)) {
+        return NULL;
+    }
+
+    if (audio_bug (AUDIO_FUNC, !drv)) {
+        dolog ("No host audio driver\n");
+        return NULL;
+    }
+
+    if (audio_bug (AUDIO_FUNC, !drv->pcm_ops)) {
+        dolog ("Host audio driver without pcm_ops\n");
+        return NULL;
+    }
+
+    hw = audio_calloc (AUDIO_FUNC, 1, glue (drv->voice_size_, TYPE));
+    if (!hw) {
+        dolog ("Can not allocate voice `%s' size %d\n",
+               drv->name, glue (drv->voice_size_, TYPE));
+        return NULL;
+    }
+
+    hw->pcm_ops = drv->pcm_ops;
+    hw->ctl_caps = drv->ctl_caps;
+
+    QLIST_INIT (&hw->sw_head);
+#ifdef DAC
+    QLIST_INIT (&hw->cap_head);
+#endif
+    if (glue (hw->pcm_ops->init_, TYPE) (hw, as, s->drv_opaque)) {
+        goto err0;
+    }
+
+    if (audio_bug (AUDIO_FUNC, hw->samples <= 0)) {
+        dolog ("hw->samples=%d\n", hw->samples);
+        goto err1;
+    }
+
+#ifdef DAC
+    hw->clip = mixeng_clip
+#else
+    hw->conv = mixeng_conv
+#endif
+        [hw->info.nchannels == 2]
+        [hw->info.sign]
+        [hw->info.swap_endianness]
+        [audio_bits_to_index (hw->info.bits)];
+
+    if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
+        goto err1;
+    }
+
+    QLIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
+    glue (s->nb_hw_voices_, TYPE) -= 1;
+#ifdef DAC
+    audio_attach_capture (hw);
+#endif
+    return hw;
+
+ err1:
+    glue (hw->pcm_ops->fini_, TYPE) (hw);
+ err0:
+    g_free (hw);
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
+{
+    HW *hw;
+
+    if (glue (conf.fixed_, TYPE).enabled && glue (conf.fixed_, TYPE).greedy) {
+        hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
+        if (hw) {
+            return hw;
+        }
+    }
+
+    hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, as);
+    if (hw) {
+        return hw;
+    }
+
+    hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
+    if (hw) {
+        return hw;
+    }
+
+    return glue (audio_pcm_hw_find_any_, TYPE) (NULL);
+}
+
+static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
+    const char *sw_name,
+    struct audsettings *as
+    )
+{
+    SW *sw;
+    HW *hw;
+    struct audsettings hw_as;
+
+    if (glue (conf.fixed_, TYPE).enabled) {
+        hw_as = glue (conf.fixed_, TYPE).settings;
+    }
+    else {
+        hw_as = *as;
+    }
+
+    sw = audio_calloc (AUDIO_FUNC, 1, sizeof (*sw));
+    if (!sw) {
+        dolog ("Could not allocate soft voice `%s' (%zu bytes)\n",
+               sw_name ? sw_name : "unknown", sizeof (*sw));
+        goto err1;
+    }
+
+    hw = glue (audio_pcm_hw_add_, TYPE) (&hw_as);
+    if (!hw) {
+        goto err2;
+    }
+
+    glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
+
+    if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
+        goto err3;
+    }
+
+    return sw;
+
+err3:
+    glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+    glue (audio_pcm_hw_gc_, TYPE) (&hw);
+err2:
+    g_free (sw);
+err1:
+    return NULL;
+}
+
+static void glue (audio_close_, TYPE) (SW *sw)
+{
+    glue (audio_pcm_sw_fini_, TYPE) (sw);
+    glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+    glue (audio_pcm_hw_gc_, TYPE) (&sw->hw);
+    g_free (sw);
+}
+
+void glue (AUD_close_, TYPE) (QEMUSoundCard *card, SW *sw)
+{
+    if (sw) {
+        if (audio_bug (AUDIO_FUNC, !card)) {
+            dolog ("card=%p\n", card);
+            return;
+        }
+
+        glue (audio_close_, TYPE) (sw);
+    }
+}
+
+SW *glue (AUD_open_, TYPE) (
+    QEMUSoundCard *card,
+    SW *sw,
+    const char *name,
+    void *callback_opaque ,
+    audio_callback_fn callback_fn,
+    struct audsettings *as
+    )
+{
+    AudioState *s = &glob_audio_state;
+
+    if (audio_bug (AUDIO_FUNC, !card || !name || !callback_fn || !as)) {
+        dolog ("card=%p name=%p callback_fn=%p as=%p\n",
+               card, name, callback_fn, as);
+        goto fail;
+    }
+
+    ldebug ("open %s, freq %d, nchannels %d, fmt %d\n",
+            name, as->freq, as->nchannels, as->fmt);
+
+    if (audio_bug (AUDIO_FUNC, audio_validate_settings (as))) {
+        audio_print_settings (as);
+        goto fail;
+    }
+
+    if (audio_bug (AUDIO_FUNC, !s->drv)) {
+        dolog ("Can not open `%s' (no host audio driver)\n", name);
+        goto fail;
+    }
+
+    if (sw && audio_pcm_info_eq (&sw->info, as)) {
+        return sw;
+    }
+
+    if (!glue (conf.fixed_, TYPE).enabled && sw) {
+        glue (AUD_close_, TYPE) (card, sw);
+        sw = NULL;
+    }
+
+    if (sw) {
+        HW *hw = sw->hw;
+
+        if (!hw) {
+            dolog ("Internal logic error voice `%s' has no hardware store\n",
+                   SW_NAME (sw));
+            goto fail;
+        }
+
+        glue (audio_pcm_sw_fini_, TYPE) (sw);
+        if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
+            goto fail;
+        }
+    }
+    else {
+        sw = glue (audio_pcm_create_voice_pair_, TYPE) (name, as);
+        if (!sw) {
+            dolog ("Failed to create voice `%s'\n", name);
+            return NULL;
+        }
+    }
+
+    sw->card = card;
+    sw->vol = nominal_volume;
+    sw->callback.fn = callback_fn;
+    sw->callback.opaque = callback_opaque;
+
+#ifdef DEBUG_AUDIO
+    dolog ("%s\n", name);
+    audio_pcm_print_info ("hw", &sw->hw->info);
+    audio_pcm_print_info ("sw", &sw->info);
+#endif
+
+    return sw;
+
+ fail:
+    glue (AUD_close_, TYPE) (card, sw);
+    return NULL;
+}
+
+int glue (AUD_is_active_, TYPE) (SW *sw)
+{
+    return sw ? sw->active : 0;
+}
+
+void glue (AUD_init_time_stamp_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
+{
+    if (!sw) {
+        return;
+    }
+
+    ts->old_ts = sw->hw->ts_helper;
+}
+
+uint64_t glue (AUD_get_elapsed_usec_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
+{
+    uint64_t delta, cur_ts, old_ts;
+
+    if (!sw) {
+        return 0;
+    }
+
+    cur_ts = sw->hw->ts_helper;
+    old_ts = ts->old_ts;
+    /* dolog ("cur %" PRId64 " old %" PRId64 "\n", cur_ts, old_ts); */
+
+    if (cur_ts >= old_ts) {
+        delta = cur_ts - old_ts;
+    }
+    else {
+        delta = UINT64_MAX - old_ts + cur_ts;
+    }
+
+    if (!delta) {
+        return 0;
+    }
+
+    return muldiv64 (delta, sw->hw->info.freq, 1000000);
+}
+
+#undef TYPE
+#undef HW
+#undef SW
+#undef HWBUF
+#undef NAME
diff --git a/qemu/audio/audio_win_int.c b/qemu/audio/audio_win_int.c
new file mode 100644
index 000000000..e1324056a
--- /dev/null
+++ b/qemu/audio/audio_win_int.c
@@ -0,0 +1,107 @@
+/* public domain */
+
+#include "qemu-common.h"
+
+#define AUDIO_CAP "win-int"
+#include <windows.h>
+#include <mmsystem.h>
+
+#include "audio.h"
+#include "audio_int.h"
+#include "audio_win_int.h"
+
+int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
+                                    struct audsettings *as)
+{
+    memset (wfx, 0, sizeof (*wfx));
+
+    wfx->wFormatTag = WAVE_FORMAT_PCM;
+    wfx->nChannels = as->nchannels;
+    wfx->nSamplesPerSec = as->freq;
+    wfx->nAvgBytesPerSec = as->freq << (as->nchannels == 2);
+    wfx->nBlockAlign = 1 << (as->nchannels == 2);
+    wfx->cbSize = 0;
+
+    switch (as->fmt) {
+    case AUD_FMT_S8:
+    case AUD_FMT_U8:
+        wfx->wBitsPerSample = 8;
+        break;
+
+    case AUD_FMT_S16:
+    case AUD_FMT_U16:
+        wfx->wBitsPerSample = 16;
+        wfx->nAvgBytesPerSec <<= 1;
+        wfx->nBlockAlign <<= 1;
+        break;
+
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        wfx->wBitsPerSample = 32;
+        wfx->nAvgBytesPerSec <<= 2;
+        wfx->nBlockAlign <<= 2;
+        break;
+
+    default:
+        dolog ("Internal logic error: Bad audio format %d\n", as->freq);
+        return -1;
+    }
+
+    return 0;
+}
+
+int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
+                                  struct audsettings *as)
+{
+    if (wfx->wFormatTag != WAVE_FORMAT_PCM) {
+        dolog ("Invalid wave format, tag is not PCM, but %d\n",
+               wfx->wFormatTag);
+        return -1;
+    }
+
+    if (!wfx->nSamplesPerSec) {
+        dolog ("Invalid wave format, frequency is zero\n");
+        return -1;
+    }
+    as->freq = wfx->nSamplesPerSec;
+
+    switch (wfx->nChannels) {
+    case 1:
+        as->nchannels = 1;
+        break;
+
+    case 2:
+        as->nchannels = 2;
+        break;
+
+    default:
+        dolog (
+            "Invalid wave format, number of channels is not 1 or 2, but %d\n",
+            wfx->nChannels
+            );
+        return -1;
+    }
+
+    switch (wfx->wBitsPerSample) {
+    case 8:
+        as->fmt = AUD_FMT_U8;
+        break;
+
+    case 16:
+        as->fmt = AUD_FMT_S16;
+        break;
+
+    case 32:
+        as->fmt = AUD_FMT_S32;
+        break;
+
+    default:
+        dolog ("Invalid wave format, bits per sample is not "
+               "8, 16 or 32, but %d\n",
+               wfx->wBitsPerSample);
+        return -1;
+    }
+
+    return 0;
+}
+
diff --git a/qemu/audio/audio_win_int.h b/qemu/audio/audio_win_int.h
new file mode 100644
index 000000000..fa5b3cb80
--- /dev/null
+++ b/qemu/audio/audio_win_int.h
@@ -0,0 +1,10 @@
+#ifndef AUDIO_WIN_INT_H
+#define AUDIO_WIN_INT_H
+
+int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
+                                    struct audsettings *as);
+
+int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
+                                  struct audsettings *as);
+
+#endif /* AUDIO_WIN_INT_H */
diff --git a/qemu/audio/coreaudio.c b/qemu/audio/coreaudio.c
new file mode 100644
index 000000000..6dfd63eb4
--- /dev/null
+++ b/qemu/audio/coreaudio.c
@@ -0,0 +1,555 @@
+/*
+ * QEMU OS X CoreAudio audio driver
+ *
+ * Copyright (c) 2005 Mike Kronenberg
+ *
+ * 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 <CoreAudio/CoreAudio.h>
+#include <string.h>             /* strerror */
+#include <pthread.h>            /* pthread_X */
+
+#include "qemu-common.h"
+#include "audio.h"
+
+#define AUDIO_CAP "coreaudio"
+#include "audio_int.h"
+
+static int isAtexit;
+
+typedef struct {
+    int buffer_frames;
+    int nbuffers;
+} CoreaudioConf;
+
+typedef struct coreaudioVoiceOut {
+    HWVoiceOut hw;
+    pthread_mutex_t mutex;
+    AudioDeviceID outputDeviceID;
+    UInt32 audioDevicePropertyBufferFrameSize;
+    AudioStreamBasicDescription outputStreamBasicDescription;
+    int live;
+    int decr;
+    int rpos;
+} coreaudioVoiceOut;
+
+static void coreaudio_logstatus (OSStatus status)
+{
+    const char *str = "BUG";
+
+    switch(status) {
+    case kAudioHardwareNoError:
+        str = "kAudioHardwareNoError";
+        break;
+
+    case kAudioHardwareNotRunningError:
+        str = "kAudioHardwareNotRunningError";
+        break;
+
+    case kAudioHardwareUnspecifiedError:
+        str = "kAudioHardwareUnspecifiedError";
+        break;
+
+    case kAudioHardwareUnknownPropertyError:
+        str = "kAudioHardwareUnknownPropertyError";
+        break;
+
+    case kAudioHardwareBadPropertySizeError:
+        str = "kAudioHardwareBadPropertySizeError";
+        break;
+
+    case kAudioHardwareIllegalOperationError:
+        str = "kAudioHardwareIllegalOperationError";
+        break;
+
+    case kAudioHardwareBadDeviceError:
+        str = "kAudioHardwareBadDeviceError";
+        break;
+
+    case kAudioHardwareBadStreamError:
+        str = "kAudioHardwareBadStreamError";
+        break;
+
+    case kAudioHardwareUnsupportedOperationError:
+        str = "kAudioHardwareUnsupportedOperationError";
+        break;
+
+    case kAudioDeviceUnsupportedFormatError:
+        str = "kAudioDeviceUnsupportedFormatError";
+        break;
+
+    case kAudioDevicePermissionsError:
+        str = "kAudioDevicePermissionsError";
+        break;
+
+    default:
+        AUD_log (AUDIO_CAP, "Reason: status code %" PRId32 "\n", (int32_t)status);
+        return;
+    }
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", str);
+}
+
+static void GCC_FMT_ATTR (2, 3) coreaudio_logerr (
+    OSStatus status,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_log (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    coreaudio_logstatus (status);
+}
+
+static void GCC_FMT_ATTR (3, 4) coreaudio_logerr2 (
+    OSStatus status,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    coreaudio_logstatus (status);
+}
+
+static inline UInt32 isPlaying (AudioDeviceID outputDeviceID)
+{
+    OSStatus status;
+    UInt32 result = 0;
+    UInt32 propertySize = sizeof(outputDeviceID);
+    status = AudioDeviceGetProperty(
+        outputDeviceID, 0, 0,
+        kAudioDevicePropertyDeviceIsRunning, &propertySize, &result);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr(status,
+                         "Could not determine whether Device is playing\n");
+    }
+    return result;
+}
+
+static void coreaudio_atexit (void)
+{
+    isAtexit = 1;
+}
+
+static int coreaudio_lock (coreaudioVoiceOut *core, const char *fn_name)
+{
+    int err;
+
+    err = pthread_mutex_lock (&core->mutex);
+    if (err) {
+        dolog ("Could not lock voice for %s\nReason: %s\n",
+               fn_name, strerror (err));
+        return -1;
+    }
+    return 0;
+}
+
+static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
+{
+    int err;
+
+    err = pthread_mutex_unlock (&core->mutex);
+    if (err) {
+        dolog ("Could not unlock voice for %s\nReason: %s\n",
+               fn_name, strerror (err));
+        return -1;
+    }
+    return 0;
+}
+
+static int coreaudio_run_out (HWVoiceOut *hw, int live)
+{
+    int decr;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    if (coreaudio_lock (core, "coreaudio_run_out")) {
+        return 0;
+    }
+
+    if (core->decr > live) {
+        ldebug ("core->decr %d live %d core->live %d\n",
+                core->decr,
+                live,
+                core->live);
+    }
+
+    decr = audio_MIN (core->decr, live);
+    core->decr -= decr;
+
+    core->live = live - decr;
+    hw->rpos = core->rpos;
+
+    coreaudio_unlock (core, "coreaudio_run_out");
+    return decr;
+}
+
+/* callback to feed audiooutput buffer */
+static OSStatus audioDeviceIOProc(
+    AudioDeviceID inDevice,
+    const AudioTimeStamp* inNow,
+    const AudioBufferList* inInputData,
+    const AudioTimeStamp* inInputTime,
+    AudioBufferList* outOutputData,
+    const AudioTimeStamp* inOutputTime,
+    void* hwptr)
+{
+    UInt32 frame, frameCount;
+    float *out = outOutputData->mBuffers[0].mData;
+    HWVoiceOut *hw = hwptr;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
+    int rpos, live;
+    struct st_sample *src;
+#ifndef FLOAT_MIXENG
+#ifdef RECIPROCAL
+    const float scale = 1.f / UINT_MAX;
+#else
+    const float scale = UINT_MAX;
+#endif
+#endif
+
+    if (coreaudio_lock (core, "audioDeviceIOProc")) {
+        inInputTime = 0;
+        return 0;
+    }
+
+    frameCount = core->audioDevicePropertyBufferFrameSize;
+    live = core->live;
+
+    /* if there are not enough samples, set signal and return */
+    if (live < frameCount) {
+        inInputTime = 0;
+        coreaudio_unlock (core, "audioDeviceIOProc(empty)");
+        return 0;
+    }
+
+    rpos = core->rpos;
+    src = hw->mix_buf + rpos;
+
+    /* fill buffer */
+    for (frame = 0; frame < frameCount; frame++) {
+#ifdef FLOAT_MIXENG
+        *out++ = src[frame].l; /* left channel */
+        *out++ = src[frame].r; /* right channel */
+#else
+#ifdef RECIPROCAL
+        *out++ = src[frame].l * scale; /* left channel */
+        *out++ = src[frame].r * scale; /* right channel */
+#else
+        *out++ = src[frame].l / scale; /* left channel */
+        *out++ = src[frame].r / scale; /* right channel */
+#endif
+#endif
+    }
+
+    rpos = (rpos + frameCount) % hw->samples;
+    core->decr += frameCount;
+    core->rpos = rpos;
+
+    coreaudio_unlock (core, "audioDeviceIOProc");
+    return 0;
+}
+
+static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
+                              void *drv_opaque)
+{
+    OSStatus status;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+    UInt32 propertySize;
+    int err;
+    const char *typ = "playback";
+    AudioValueRange frameRange;
+    CoreaudioConf *conf = drv_opaque;
+
+    /* create mutex */
+    err = pthread_mutex_init(&core->mutex, NULL);
+    if (err) {
+        dolog("Could not create mutex\nReason: %s\n", strerror (err));
+        return -1;
+    }
+
+    audio_pcm_init_info (&hw->info, as);
+
+    /* open default output device */
+    propertySize = sizeof(core->outputDeviceID);
+    status = AudioHardwareGetProperty(
+        kAudioHardwarePropertyDefaultOutputDevice,
+        &propertySize,
+        &core->outputDeviceID);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Could not get default output Device\n");
+        return -1;
+    }
+    if (core->outputDeviceID == kAudioDeviceUnknown) {
+        dolog ("Could not initialize %s - Unknown Audiodevice\n", typ);
+        return -1;
+    }
+
+    /* get minimum and maximum buffer frame sizes */
+    propertySize = sizeof(frameRange);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        0,
+        kAudioDevicePropertyBufferFrameSizeRange,
+        &propertySize,
+        &frameRange);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Could not get device buffer frame range\n");
+        return -1;
+    }
+
+    if (frameRange.mMinimum > conf->buffer_frames) {
+        core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
+        dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
+    }
+    else if (frameRange.mMaximum < conf->buffer_frames) {
+        core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
+        dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
+    }
+    else {
+        core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
+    }
+
+    /* set Buffer Frame Size */
+    propertySize = sizeof(core->audioDevicePropertyBufferFrameSize);
+    status = AudioDeviceSetProperty(
+        core->outputDeviceID,
+        NULL,
+        0,
+        false,
+        kAudioDevicePropertyBufferFrameSize,
+        propertySize,
+        &core->audioDevicePropertyBufferFrameSize);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Could not set device buffer frame size %" PRIu32 "\n",
+                           (uint32_t)core->audioDevicePropertyBufferFrameSize);
+        return -1;
+    }
+
+    /* get Buffer Frame Size */
+    propertySize = sizeof(core->audioDevicePropertyBufferFrameSize);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        false,
+        kAudioDevicePropertyBufferFrameSize,
+        &propertySize,
+        &core->audioDevicePropertyBufferFrameSize);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Could not get device buffer frame size\n");
+        return -1;
+    }
+    hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
+
+    /* get StreamFormat */
+    propertySize = sizeof(core->outputStreamBasicDescription);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        false,
+        kAudioDevicePropertyStreamFormat,
+        &propertySize,
+        &core->outputStreamBasicDescription);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Could not get Device Stream properties\n");
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* set Samplerate */
+    core->outputStreamBasicDescription.mSampleRate = (Float64) as->freq;
+    propertySize = sizeof(core->outputStreamBasicDescription);
+    status = AudioDeviceSetProperty(
+        core->outputDeviceID,
+        0,
+        0,
+        0,
+        kAudioDevicePropertyStreamFormat,
+        propertySize,
+        &core->outputStreamBasicDescription);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Could not set samplerate %d\n",
+                           as->freq);
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* set Callback */
+    status = AudioDeviceAddIOProc(core->outputDeviceID, audioDeviceIOProc, hw);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Could not set IOProc\n");
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* start Playback */
+    if (!isPlaying(core->outputDeviceID)) {
+        status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+        if (status != kAudioHardwareNoError) {
+            coreaudio_logerr2 (status, typ, "Could not start playback\n");
+            AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+            core->outputDeviceID = kAudioDeviceUnknown;
+            return -1;
+        }
+    }
+
+    return 0;
+}
+
+static void coreaudio_fini_out (HWVoiceOut *hw)
+{
+    OSStatus status;
+    int err;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    if (!isAtexit) {
+        /* stop playback */
+        if (isPlaying(core->outputDeviceID)) {
+            status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+            if (status != kAudioHardwareNoError) {
+                coreaudio_logerr (status, "Could not stop playback\n");
+            }
+        }
+
+        /* remove callback */
+        status = AudioDeviceRemoveIOProc(core->outputDeviceID,
+                                         audioDeviceIOProc);
+        if (status != kAudioHardwareNoError) {
+            coreaudio_logerr (status, "Could not remove IOProc\n");
+        }
+    }
+    core->outputDeviceID = kAudioDeviceUnknown;
+
+    /* destroy mutex */
+    err = pthread_mutex_destroy(&core->mutex);
+    if (err) {
+        dolog("Could not destroy mutex\nReason: %s\n", strerror (err));
+    }
+}
+
+static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    OSStatus status;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        /* start playback */
+        if (!isPlaying(core->outputDeviceID)) {
+            status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+            if (status != kAudioHardwareNoError) {
+                coreaudio_logerr (status, "Could not resume playback\n");
+            }
+        }
+        break;
+
+    case VOICE_DISABLE:
+        /* stop playback */
+        if (!isAtexit) {
+            if (isPlaying(core->outputDeviceID)) {
+                status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+                if (status != kAudioHardwareNoError) {
+                    coreaudio_logerr (status, "Could not pause playback\n");
+                }
+            }
+        }
+        break;
+    }
+    return 0;
+}
+
+static CoreaudioConf glob_conf = {
+    .buffer_frames = 512,
+    .nbuffers = 4,
+};
+
+static void *coreaudio_audio_init (void)
+{
+    CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
+    *conf = glob_conf;
+
+    atexit(coreaudio_atexit);
+    return conf;
+}
+
+static void coreaudio_audio_fini (void *opaque)
+{
+    g_free(opaque);
+}
+
+static struct audio_option coreaudio_options[] = {
+    {
+        .name  = "BUFFER_SIZE",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.buffer_frames,
+        .descr = "Size of the buffer in frames"
+    },
+    {
+        .name  = "BUFFER_COUNT",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.nbuffers,
+        .descr = "Number of buffers"
+    },
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops coreaudio_pcm_ops = {
+    .init_out = coreaudio_init_out,
+    .fini_out = coreaudio_fini_out,
+    .run_out  = coreaudio_run_out,
+    .write    = coreaudio_write,
+    .ctl_out  = coreaudio_ctl_out
+};
+
+struct audio_driver coreaudio_audio_driver = {
+    .name           = "coreaudio",
+    .descr          = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
+    .options        = coreaudio_options,
+    .init           = coreaudio_audio_init,
+    .fini           = coreaudio_audio_fini,
+    .pcm_ops        = &coreaudio_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = 1,
+    .max_voices_in  = 0,
+    .voice_size_out = sizeof (coreaudioVoiceOut),
+    .voice_size_in  = 0
+};
diff --git a/qemu/audio/dsound_template.h b/qemu/audio/dsound_template.h
new file mode 100644
index 000000000..b439f33f5
--- /dev/null
+++ b/qemu/audio/dsound_template.h
@@ -0,0 +1,278 @@
+/*
+ * QEMU DirectSound audio driver header
+ *
+ * Copyright (c) 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.
+ */
+#ifdef DSBTYPE_IN
+#define NAME "capture buffer"
+#define NAME2 "DirectSoundCapture"
+#define TYPE in
+#define IFACE IDirectSoundCaptureBuffer
+#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
+#define FIELD dsound_capture_buffer
+#define FIELD2 dsound_capture
+#else
+#define NAME "playback buffer"
+#define NAME2 "DirectSound"
+#define TYPE out
+#define IFACE IDirectSoundBuffer
+#define BUFPTR LPDIRECTSOUNDBUFFER
+#define FIELD dsound_buffer
+#define FIELD2 dsound
+#endif
+
+static int glue (dsound_unlock_, TYPE) (
+    BUFPTR buf,
+    LPVOID p1,
+    LPVOID p2,
+    DWORD blen1,
+    DWORD blen2
+    )
+{
+    HRESULT hr;
+
+    hr = glue (IFACE, _Unlock) (buf, p1, blen1, p2, blen2);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not unlock " NAME "\n");
+        return -1;
+    }
+
+    return 0;
+}
+
+static int glue (dsound_lock_, TYPE) (
+    BUFPTR buf,
+    struct audio_pcm_info *info,
+    DWORD pos,
+    DWORD len,
+    LPVOID *p1p,
+    LPVOID *p2p,
+    DWORD *blen1p,
+    DWORD *blen2p,
+    int entire,
+    dsound *s
+    )
+{
+    HRESULT hr;
+    LPVOID p1 = NULL, p2 = NULL;
+    DWORD blen1 = 0, blen2 = 0;
+    DWORD flag;
+
+#ifdef DSBTYPE_IN
+    flag = entire ? DSCBLOCK_ENTIREBUFFER : 0;
+#else
+    flag = entire ? DSBLOCK_ENTIREBUFFER : 0;
+#endif
+    hr = glue(IFACE, _Lock)(buf, pos, len, &p1, &blen1, &p2, &blen2, flag);
+
+    if (FAILED (hr)) {
+#ifndef DSBTYPE_IN
+        if (hr == DSERR_BUFFERLOST) {
+            if (glue (dsound_restore_, TYPE) (buf, s)) {
+                dsound_logerr (hr, "Could not lock " NAME "\n");
+            }
+            goto fail;
+        }
+#endif
+        dsound_logerr (hr, "Could not lock " NAME "\n");
+        goto fail;
+    }
+
+    if ((p1 && (blen1 & info->align)) || (p2 && (blen2 & info->align))) {
+        dolog ("DirectSound returned misaligned buffer %ld %ld\n",
+               blen1, blen2);
+        glue (dsound_unlock_, TYPE) (buf, p1, p2, blen1, blen2);
+        goto fail;
+    }
+
+    if (!p1 && blen1) {
+        dolog ("warning: !p1 && blen1=%ld\n", blen1);
+        blen1 = 0;
+    }
+
+    if (!p2 && blen2) {
+        dolog ("warning: !p2 && blen2=%ld\n", blen2);
+        blen2 = 0;
+    }
+
+    *p1p = p1;
+    *p2p = p2;
+    *blen1p = blen1;
+    *blen2p = blen2;
+    return 0;
+
+ fail:
+    *p1p = NULL - 1;
+    *p2p = NULL - 1;
+    *blen1p = -1;
+    *blen2p = -1;
+    return -1;
+}
+
+#ifdef DSBTYPE_IN
+static void dsound_fini_in (HWVoiceIn *hw)
+#else
+static void dsound_fini_out (HWVoiceOut *hw)
+#endif
+{
+    HRESULT hr;
+#ifdef DSBTYPE_IN
+    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+#else
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+#endif
+
+    if (ds->FIELD) {
+        hr = glue (IFACE, _Stop) (ds->FIELD);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Could not stop " NAME "\n");
+        }
+
+        hr = glue (IFACE, _Release) (ds->FIELD);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Could not release " NAME "\n");
+        }
+        ds->FIELD = NULL;
+    }
+}
+
+#ifdef DSBTYPE_IN
+static int dsound_init_in(HWVoiceIn *hw, struct audsettings *as,
+                          void *drv_opaque)
+#else
+static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
+                           void *drv_opaque)
+#endif
+{
+    int err;
+    HRESULT hr;
+    dsound *s = drv_opaque;
+    WAVEFORMATEX wfx;
+    struct audsettings obt_as;
+    DSoundConf *conf = &s->conf;
+#ifdef DSBTYPE_IN
+    const char *typ = "ADC";
+    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+    DSCBUFFERDESC bd;
+    DSCBCAPS bc;
+#else
+    const char *typ = "DAC";
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+    DSBUFFERDESC bd;
+    DSBCAPS bc;
+#endif
+
+    if (!s->FIELD2) {
+        dolog ("Attempt to initialize voice without " NAME2 " object\n");
+        return -1;
+    }
+
+    err = waveformat_from_audio_settings (&wfx, as);
+    if (err) {
+        return -1;
+    }
+
+    memset (&bd, 0, sizeof (bd));
+    bd.dwSize = sizeof (bd);
+    bd.lpwfxFormat = &wfx;
+#ifdef DSBTYPE_IN
+    bd.dwBufferBytes = conf->bufsize_in;
+    hr = IDirectSoundCapture_CreateCaptureBuffer (
+        s->dsound_capture,
+        &bd,
+        &ds->dsound_capture_buffer,
+        NULL
+        );
+#else
+    bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
+    bd.dwBufferBytes = conf->bufsize_out;
+    hr = IDirectSound_CreateSoundBuffer (
+        s->dsound,
+        &bd,
+        &ds->dsound_buffer,
+        NULL
+        );
+#endif
+
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Could not create " NAME "\n");
+        return -1;
+    }
+
+    hr = glue (IFACE, _GetFormat) (ds->FIELD, &wfx, sizeof (wfx), NULL);
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Could not get " NAME " format\n");
+        goto fail0;
+    }
+
+#ifdef DEBUG_DSOUND
+    dolog (NAME "\n");
+    print_wave_format (&wfx);
+#endif
+
+    memset (&bc, 0, sizeof (bc));
+    bc.dwSize = sizeof (bc);
+
+    hr = glue (IFACE, _GetCaps) (ds->FIELD, &bc);
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Could not get " NAME " format\n");
+        goto fail0;
+    }
+
+    err = waveformat_to_audio_settings (&wfx, &obt_as);
+    if (err) {
+        goto fail0;
+    }
+
+    ds->first_time = 1;
+    obt_as.endianness = 0;
+    audio_pcm_init_info (&hw->info, &obt_as);
+
+    if (bc.dwBufferBytes & hw->info.align) {
+        dolog (
+            "GetCaps returned misaligned buffer size %ld, alignment %d\n",
+            bc.dwBufferBytes, hw->info.align + 1
+            );
+    }
+    hw->samples = bc.dwBufferBytes >> hw->info.shift;
+    ds->s = s;
+
+#ifdef DEBUG_DSOUND
+    dolog ("caps %ld, desc %ld\n",
+           bc.dwBufferBytes, bd.dwBufferBytes);
+
+    dolog ("bufsize %d, freq %d, chan %d, fmt %d\n",
+           hw->bufsize, settings.freq, settings.nchannels, settings.fmt);
+#endif
+    return 0;
+
+ fail0:
+    glue (dsound_fini_, TYPE) (hw);
+    return -1;
+}
+
+#undef NAME
+#undef NAME2
+#undef TYPE
+#undef IFACE
+#undef BUFPTR
+#undef FIELD
+#undef FIELD2
diff --git a/qemu/audio/dsoundaudio.c b/qemu/audio/dsoundaudio.c
new file mode 100644
index 000000000..e9472c105
--- /dev/null
+++ b/qemu/audio/dsoundaudio.c
@@ -0,0 +1,904 @@
+/*
+ * QEMU DirectSound audio driver
+ *
+ * Copyright (c) 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.
+ */
+
+/*
+ * SEAL 1.07 by Carlos 'pel' Hasan was used as documentation
+ */
+
+#include "qemu-common.h"
+#include "audio.h"
+
+#define AUDIO_CAP "dsound"
+#include "audio_int.h"
+
+#include <windows.h>
+#include <mmsystem.h>
+#include <objbase.h>
+#include <dsound.h>
+
+#include "audio_win_int.h"
+
+/* #define DEBUG_DSOUND */
+
+typedef struct {
+    int bufsize_in;
+    int bufsize_out;
+    int latency_millis;
+} DSoundConf;
+
+typedef struct {
+    LPDIRECTSOUND dsound;
+    LPDIRECTSOUNDCAPTURE dsound_capture;
+    struct audsettings settings;
+    DSoundConf conf;
+} dsound;
+
+typedef struct {
+    HWVoiceOut hw;
+    LPDIRECTSOUNDBUFFER dsound_buffer;
+    DWORD old_pos;
+    int first_time;
+    dsound *s;
+#ifdef DEBUG_DSOUND
+    DWORD old_ppos;
+    DWORD played;
+    DWORD mixed;
+#endif
+} DSoundVoiceOut;
+
+typedef struct {
+    HWVoiceIn hw;
+    int first_time;
+    LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
+    dsound *s;
+} DSoundVoiceIn;
+
+static void dsound_log_hresult (HRESULT hr)
+{
+    const char *str = "BUG";
+
+    switch (hr) {
+    case DS_OK:
+        str = "The method succeeded";
+        break;
+#ifdef DS_NO_VIRTUALIZATION
+    case DS_NO_VIRTUALIZATION:
+        str = "The buffer was created, but another 3D algorithm was substituted";
+        break;
+#endif
+#ifdef DS_INCOMPLETE
+    case DS_INCOMPLETE:
+        str = "The method succeeded, but not all the optional effects were obtained";
+        break;
+#endif
+#ifdef DSERR_ACCESSDENIED
+    case DSERR_ACCESSDENIED:
+        str = "The request failed because access was denied";
+        break;
+#endif
+#ifdef DSERR_ALLOCATED
+    case DSERR_ALLOCATED:
+        str = "The request failed because resources, such as a priority level, were already in use by another caller";
+        break;
+#endif
+#ifdef DSERR_ALREADYINITIALIZED
+    case DSERR_ALREADYINITIALIZED:
+        str = "The object is already initialized";
+        break;
+#endif
+#ifdef DSERR_BADFORMAT
+    case DSERR_BADFORMAT:
+        str = "The specified wave format is not supported";
+        break;
+#endif
+#ifdef DSERR_BADSENDBUFFERGUID
+    case DSERR_BADSENDBUFFERGUID:
+        str = "The GUID specified in an audiopath file does not match a valid mix-in buffer";
+        break;
+#endif
+#ifdef DSERR_BUFFERLOST
+    case DSERR_BUFFERLOST:
+        str = "The buffer memory has been lost and must be restored";
+        break;
+#endif
+#ifdef DSERR_BUFFERTOOSMALL
+    case DSERR_BUFFERTOOSMALL:
+        str = "The buffer size is not great enough to enable effects processing";
+        break;
+#endif
+#ifdef DSERR_CONTROLUNAVAIL
+    case DSERR_CONTROLUNAVAIL:
+        str = "The buffer control (volume, pan, and so on) requested by the caller is not available. Controls must be specified when the buffer is created, using the dwFlags member of DSBUFFERDESC";
+        break;
+#endif
+#ifdef DSERR_DS8_REQUIRED
+    case DSERR_DS8_REQUIRED:
+        str = "A DirectSound object of class CLSID_DirectSound8 or later is required for the requested functionality. For more information, see IDirectSound8 Interface";
+        break;
+#endif
+#ifdef DSERR_FXUNAVAILABLE
+    case DSERR_FXUNAVAILABLE:
+        str = "The effects requested could not be found on the system, or they are in the wrong order or in the wrong location; for example, an effect expected in hardware was found in software";
+        break;
+#endif
+#ifdef DSERR_GENERIC
+    case DSERR_GENERIC :
+        str = "An undetermined error occurred inside the DirectSound subsystem";
+        break;
+#endif
+#ifdef DSERR_INVALIDCALL
+    case DSERR_INVALIDCALL:
+        str = "This function is not valid for the current state of this object";
+        break;
+#endif
+#ifdef DSERR_INVALIDPARAM
+    case DSERR_INVALIDPARAM:
+        str = "An invalid parameter was passed to the returning function";
+        break;
+#endif
+#ifdef DSERR_NOAGGREGATION
+    case DSERR_NOAGGREGATION:
+        str = "The object does not support aggregation";
+        break;
+#endif
+#ifdef DSERR_NODRIVER
+    case DSERR_NODRIVER:
+        str = "No sound driver is available for use, or the given GUID is not a valid DirectSound device ID";
+        break;
+#endif
+#ifdef DSERR_NOINTERFACE
+    case DSERR_NOINTERFACE:
+        str = "The requested COM interface is not available";
+        break;
+#endif
+#ifdef DSERR_OBJECTNOTFOUND
+    case DSERR_OBJECTNOTFOUND:
+        str = "The requested object was not found";
+        break;
+#endif
+#ifdef DSERR_OTHERAPPHASPRIO
+    case DSERR_OTHERAPPHASPRIO:
+        str = "Another application has a higher priority level, preventing this call from succeeding";
+        break;
+#endif
+#ifdef DSERR_OUTOFMEMORY
+    case DSERR_OUTOFMEMORY:
+        str = "The DirectSound subsystem could not allocate sufficient memory to complete the caller's request";
+        break;
+#endif
+#ifdef DSERR_PRIOLEVELNEEDED
+    case DSERR_PRIOLEVELNEEDED:
+        str = "A cooperative level of DSSCL_PRIORITY or higher is required";
+        break;
+#endif
+#ifdef DSERR_SENDLOOP
+    case DSERR_SENDLOOP:
+        str = "A circular loop of send effects was detected";
+        break;
+#endif
+#ifdef DSERR_UNINITIALIZED
+    case DSERR_UNINITIALIZED:
+        str = "The Initialize method has not been called or has not been called successfully before other methods were called";
+        break;
+#endif
+#ifdef DSERR_UNSUPPORTED
+    case DSERR_UNSUPPORTED:
+        str = "The function called is not supported at this time";
+        break;
+#endif
+    default:
+        AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT %#lx)\n", hr);
+        return;
+    }
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", str);
+}
+
+static void GCC_FMT_ATTR (2, 3) dsound_logerr (
+    HRESULT hr,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    dsound_log_hresult (hr);
+}
+
+static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
+    HRESULT hr,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    dsound_log_hresult (hr);
+}
+
+static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
+{
+    return (millis * info->bytes_per_second) / 1000;
+}
+
+#ifdef DEBUG_DSOUND
+static void print_wave_format (WAVEFORMATEX *wfx)
+{
+    dolog ("tag             = %d\n", wfx->wFormatTag);
+    dolog ("nChannels       = %d\n", wfx->nChannels);
+    dolog ("nSamplesPerSec  = %ld\n", wfx->nSamplesPerSec);
+    dolog ("nAvgBytesPerSec = %ld\n", wfx->nAvgBytesPerSec);
+    dolog ("nBlockAlign     = %d\n", wfx->nBlockAlign);
+    dolog ("wBitsPerSample  = %d\n", wfx->wBitsPerSample);
+    dolog ("cbSize          = %d\n", wfx->cbSize);
+}
+#endif
+
+static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb, dsound *s)
+{
+    HRESULT hr;
+
+    hr = IDirectSoundBuffer_Restore (dsb);
+
+    if (hr != DS_OK) {
+        dsound_logerr (hr, "Could not restore playback buffer\n");
+        return -1;
+    }
+    return 0;
+}
+
+#include "dsound_template.h"
+#define DSBTYPE_IN
+#include "dsound_template.h"
+#undef DSBTYPE_IN
+
+static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp,
+                                  dsound *s)
+{
+    HRESULT hr;
+
+    hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not get playback buffer status\n");
+        return -1;
+    }
+
+    if (*statusp & DSERR_BUFFERLOST) {
+        dsound_restore_out(dsb, s);
+        return -1;
+    }
+
+    return 0;
+}
+
+static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,
+                                 DWORD *statusp)
+{
+    HRESULT hr;
+
+    hr = IDirectSoundCaptureBuffer_GetStatus (dscb, statusp);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not get capture buffer status\n");
+        return -1;
+    }
+
+    return 0;
+}
+
+static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
+{
+    int src_len1 = dst_len;
+    int src_len2 = 0;
+    int pos = hw->rpos + dst_len;
+    struct st_sample *src1 = hw->mix_buf + hw->rpos;
+    struct st_sample *src2 = NULL;
+
+    if (pos > hw->samples) {
+        src_len1 = hw->samples - hw->rpos;
+        src2 = hw->mix_buf;
+        src_len2 = dst_len - src_len1;
+        pos = src_len2;
+    }
+
+    if (src_len1) {
+        hw->clip (dst, src1, src_len1);
+    }
+
+    if (src_len2) {
+        dst = advance (dst, src_len1 << hw->info.shift);
+        hw->clip (dst, src2, src_len2);
+    }
+
+    hw->rpos = pos % hw->samples;
+}
+
+static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
+                                 dsound *s)
+{
+    int err;
+    LPVOID p1, p2;
+    DWORD blen1, blen2, len1, len2;
+
+    err = dsound_lock_out (
+        dsb,
+        &hw->info,
+        0,
+        hw->samples << hw->info.shift,
+        &p1, &p2,
+        &blen1, &blen2,
+        1,
+        s
+        );
+    if (err) {
+        return;
+    }
+
+    len1 = blen1 >> hw->info.shift;
+    len2 = blen2 >> hw->info.shift;
+
+#ifdef DEBUG_DSOUND
+    dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
+           p1, blen1, len1,
+           p2, blen2, len2);
+#endif
+
+    if (p1 && len1) {
+        audio_pcm_info_clear_buf (&hw->info, p1, len1);
+    }
+
+    if (p2 && len2) {
+        audio_pcm_info_clear_buf (&hw->info, p2, len2);
+    }
+
+    dsound_unlock_out (dsb, p1, p2, blen1, blen2);
+}
+
+static int dsound_open (dsound *s)
+{
+    HRESULT hr;
+    HWND hwnd;
+
+    hwnd = GetForegroundWindow ();
+    hr = IDirectSound_SetCooperativeLevel (
+        s->dsound,
+        hwnd,
+        DSSCL_PRIORITY
+        );
+
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not set cooperative level for window %p\n",
+                       hwnd);
+        return -1;
+    }
+
+    return 0;
+}
+
+static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    HRESULT hr;
+    DWORD status;
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+    LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+    dsound *s = ds->s;
+
+    if (!dsb) {
+        dolog ("Attempt to control voice without a buffer\n");
+        return 0;
+    }
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        if (dsound_get_status_out (dsb, &status, s)) {
+            return -1;
+        }
+
+        if (status & DSBSTATUS_PLAYING) {
+            dolog ("warning: Voice is already playing\n");
+            return 0;
+        }
+
+        dsound_clear_sample (hw, dsb, s);
+
+        hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Could not start playing buffer\n");
+            return -1;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        if (dsound_get_status_out (dsb, &status, s)) {
+            return -1;
+        }
+
+        if (status & DSBSTATUS_PLAYING) {
+            hr = IDirectSoundBuffer_Stop (dsb);
+            if (FAILED (hr)) {
+                dsound_logerr (hr, "Could not stop playing buffer\n");
+                return -1;
+            }
+        }
+        else {
+            dolog ("warning: Voice is not playing\n");
+        }
+        break;
+    }
+    return 0;
+}
+
+static int dsound_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int dsound_run_out (HWVoiceOut *hw, int live)
+{
+    int err;
+    HRESULT hr;
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+    LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+    int len, hwshift;
+    DWORD blen1, blen2;
+    DWORD len1, len2;
+    DWORD decr;
+    DWORD wpos, ppos, old_pos;
+    LPVOID p1, p2;
+    int bufsize;
+    dsound *s = ds->s;
+    DSoundConf *conf = &s->conf;
+
+    if (!dsb) {
+        dolog ("Attempt to run empty with playback buffer\n");
+        return 0;
+    }
+
+    hwshift = hw->info.shift;
+    bufsize = hw->samples << hwshift;
+
+    hr = IDirectSoundBuffer_GetCurrentPosition (
+        dsb,
+        &ppos,
+        ds->first_time ? &wpos : NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not get playback buffer position\n");
+        return 0;
+    }
+
+    len = live << hwshift;
+
+    if (ds->first_time) {
+        if (conf->latency_millis) {
+            DWORD cur_blat;
+
+            cur_blat = audio_ring_dist (wpos, ppos, bufsize);
+            ds->first_time = 0;
+            old_pos = wpos;
+            old_pos +=
+                millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
+            old_pos %= bufsize;
+            old_pos &= ~hw->info.align;
+        }
+        else {
+            old_pos = wpos;
+        }
+#ifdef DEBUG_DSOUND
+        ds->played = 0;
+        ds->mixed = 0;
+#endif
+    }
+    else {
+        if (ds->old_pos == ppos) {
+#ifdef DEBUG_DSOUND
+            dolog ("old_pos == ppos\n");
+#endif
+            return 0;
+        }
+
+#ifdef DEBUG_DSOUND
+        ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
+#endif
+        old_pos = ds->old_pos;
+    }
+
+    if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
+        len = ppos - old_pos;
+    }
+    else {
+        if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
+            len = bufsize - old_pos + ppos;
+        }
+    }
+
+    if (audio_bug (AUDIO_FUNC, len < 0 || len > bufsize)) {
+        dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n",
+               len, bufsize, old_pos, ppos);
+        return 0;
+    }
+
+    len &= ~hw->info.align;
+    if (!len) {
+        return 0;
+    }
+
+#ifdef DEBUG_DSOUND
+    ds->old_ppos = ppos;
+#endif
+    err = dsound_lock_out (
+        dsb,
+        &hw->info,
+        old_pos,
+        len,
+        &p1, &p2,
+        &blen1, &blen2,
+        0,
+        s
+        );
+    if (err) {
+        return 0;
+    }
+
+    len1 = blen1 >> hwshift;
+    len2 = blen2 >> hwshift;
+    decr = len1 + len2;
+
+    if (p1 && len1) {
+        dsound_write_sample (hw, p1, len1);
+    }
+
+    if (p2 && len2) {
+        dsound_write_sample (hw, p2, len2);
+    }
+
+    dsound_unlock_out (dsb, p1, p2, blen1, blen2);
+    ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;
+
+#ifdef DEBUG_DSOUND
+    ds->mixed += decr << hwshift;
+
+    dolog ("played %lu mixed %lu diff %ld sec %f\n",
+           ds->played,
+           ds->mixed,
+           ds->mixed - ds->played,
+           abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
+#endif
+    return decr;
+}
+
+static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    HRESULT hr;
+    DWORD status;
+    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+    LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
+
+    if (!dscb) {
+        dolog ("Attempt to control capture voice without a buffer\n");
+        return -1;
+    }
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        if (dsound_get_status_in (dscb, &status)) {
+            return -1;
+        }
+
+        if (status & DSCBSTATUS_CAPTURING) {
+            dolog ("warning: Voice is already capturing\n");
+            return 0;
+        }
+
+        /* clear ?? */
+
+        hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Could not start capturing\n");
+            return -1;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        if (dsound_get_status_in (dscb, &status)) {
+            return -1;
+        }
+
+        if (status & DSCBSTATUS_CAPTURING) {
+            hr = IDirectSoundCaptureBuffer_Stop (dscb);
+            if (FAILED (hr)) {
+                dsound_logerr (hr, "Could not stop capturing\n");
+                return -1;
+            }
+        }
+        else {
+            dolog ("warning: Voice is not capturing\n");
+        }
+        break;
+    }
+    return 0;
+}
+
+static int dsound_read (SWVoiceIn *sw, void *buf, int len)
+{
+    return audio_pcm_sw_read (sw, buf, len);
+}
+
+static int dsound_run_in (HWVoiceIn *hw)
+{
+    int err;
+    HRESULT hr;
+    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+    LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
+    int live, len, dead;
+    DWORD blen1, blen2;
+    DWORD len1, len2;
+    DWORD decr;
+    DWORD cpos, rpos;
+    LPVOID p1, p2;
+    int hwshift;
+    dsound *s = ds->s;
+
+    if (!dscb) {
+        dolog ("Attempt to run without capture buffer\n");
+        return 0;
+    }
+
+    hwshift = hw->info.shift;
+
+    live = audio_pcm_hw_get_live_in (hw);
+    dead = hw->samples - live;
+    if (!dead) {
+        return 0;
+    }
+
+    hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
+        dscb,
+        &cpos,
+        ds->first_time ? &rpos : NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not get capture buffer position\n");
+        return 0;
+    }
+
+    if (ds->first_time) {
+        ds->first_time = 0;
+        if (rpos & hw->info.align) {
+            ldebug ("warning: Misaligned capture read position %ld(%d)\n",
+                    rpos, hw->info.align);
+        }
+        hw->wpos = rpos >> hwshift;
+    }
+
+    if (cpos & hw->info.align) {
+        ldebug ("warning: Misaligned capture position %ld(%d)\n",
+                cpos, hw->info.align);
+    }
+    cpos >>= hwshift;
+
+    len = audio_ring_dist (cpos, hw->wpos, hw->samples);
+    if (!len) {
+        return 0;
+    }
+    len = audio_MIN (len, dead);
+
+    err = dsound_lock_in (
+        dscb,
+        &hw->info,
+        hw->wpos << hwshift,
+        len << hwshift,
+        &p1,
+        &p2,
+        &blen1,
+        &blen2,
+        0,
+        s
+        );
+    if (err) {
+        return 0;
+    }
+
+    len1 = blen1 >> hwshift;
+    len2 = blen2 >> hwshift;
+    decr = len1 + len2;
+
+    if (p1 && len1) {
+        hw->conv (hw->conv_buf + hw->wpos, p1, len1);
+    }
+
+    if (p2 && len2) {
+        hw->conv (hw->conv_buf, p2, len2);
+    }
+
+    dsound_unlock_in (dscb, p1, p2, blen1, blen2);
+    hw->wpos = (hw->wpos + decr) % hw->samples;
+    return decr;
+}
+
+static DSoundConf glob_conf = {
+    .bufsize_in         = 16384,
+    .bufsize_out        = 16384,
+    .latency_millis     = 10
+};
+
+static void dsound_audio_fini (void *opaque)
+{
+    HRESULT hr;
+    dsound *s = opaque;
+
+    if (!s->dsound) {
+        g_free(s);
+        return;
+    }
+
+    hr = IDirectSound_Release (s->dsound);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not release DirectSound\n");
+    }
+    s->dsound = NULL;
+
+    if (!s->dsound_capture) {
+        g_free(s);
+        return;
+    }
+
+    hr = IDirectSoundCapture_Release (s->dsound_capture);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not release DirectSoundCapture\n");
+    }
+    s->dsound_capture = NULL;
+
+    g_free(s);
+}
+
+static void *dsound_audio_init (void)
+{
+    int err;
+    HRESULT hr;
+    dsound *s = g_malloc0(sizeof(dsound));
+
+    s->conf = glob_conf;
+    hr = CoInitialize (NULL);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not initialize COM\n");
+        g_free(s);
+        return NULL;
+    }
+
+    hr = CoCreateInstance (
+        &CLSID_DirectSound,
+        NULL,
+        CLSCTX_ALL,
+        &IID_IDirectSound,
+        (void **) &s->dsound
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not create DirectSound instance\n");
+        g_free(s);
+        return NULL;
+    }
+
+    hr = IDirectSound_Initialize (s->dsound, NULL);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not initialize DirectSound\n");
+
+        hr = IDirectSound_Release (s->dsound);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Could not release DirectSound\n");
+        }
+        g_free(s);
+        return NULL;
+    }
+
+    hr = CoCreateInstance (
+        &CLSID_DirectSoundCapture,
+        NULL,
+        CLSCTX_ALL,
+        &IID_IDirectSoundCapture,
+        (void **) &s->dsound_capture
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Could not create DirectSoundCapture instance\n");
+    }
+    else {
+        hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Could not initialize DirectSoundCapture\n");
+
+            hr = IDirectSoundCapture_Release (s->dsound_capture);
+            if (FAILED (hr)) {
+                dsound_logerr (hr, "Could not release DirectSoundCapture\n");
+            }
+            s->dsound_capture = NULL;
+        }
+    }
+
+    err = dsound_open (s);
+    if (err) {
+        dsound_audio_fini (s);
+        return NULL;
+    }
+
+    return s;
+}
+
+static struct audio_option dsound_options[] = {
+    {
+        .name  = "LATENCY_MILLIS",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.latency_millis,
+        .descr = "(undocumented)"
+    },
+    {
+        .name  = "BUFSIZE_OUT",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.bufsize_out,
+        .descr = "(undocumented)"
+    },
+    {
+        .name  = "BUFSIZE_IN",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.bufsize_in,
+        .descr = "(undocumented)"
+    },
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops dsound_pcm_ops = {
+    .init_out = dsound_init_out,
+    .fini_out = dsound_fini_out,
+    .run_out  = dsound_run_out,
+    .write    = dsound_write,
+    .ctl_out  = dsound_ctl_out,
+
+    .init_in  = dsound_init_in,
+    .fini_in  = dsound_fini_in,
+    .run_in   = dsound_run_in,
+    .read     = dsound_read,
+    .ctl_in   = dsound_ctl_in
+};
+
+struct audio_driver dsound_audio_driver = {
+    .name           = "dsound",
+    .descr          = "DirectSound http://wikipedia.org/wiki/DirectSound",
+    .options        = dsound_options,
+    .init           = dsound_audio_init,
+    .fini           = dsound_audio_fini,
+    .pcm_ops        = &dsound_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = 1,
+    .voice_size_out = sizeof (DSoundVoiceOut),
+    .voice_size_in  = sizeof (DSoundVoiceIn)
+};
diff --git a/qemu/audio/mixeng.c b/qemu/audio/mixeng.c
new file mode 100644
index 000000000..0e4976f27
--- /dev/null
+++ b/qemu/audio/mixeng.c
@@ -0,0 +1,366 @@
+/*
+ * QEMU Mixing engine
+ *
+ * Copyright (c) 2004-2005 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 "qemu-common.h"
+#include "audio.h"
+
+#define AUDIO_CAP "mixeng"
+#include "audio_int.h"
+
+/* 8 bit */
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+
+/* Signed 8 bit */
+#define BSIZE 8
+#define ITYPE int
+#define IN_MIN SCHAR_MIN
+#define IN_MAX SCHAR_MAX
+#define SIGNED
+#define SHIFT 8
+#include "mixeng_template.h"
+#undef SIGNED
+#undef IN_MAX
+#undef IN_MIN
+#undef BSIZE
+#undef ITYPE
+#undef SHIFT
+
+/* Unsigned 8 bit */
+#define BSIZE 8
+#define ITYPE uint
+#define IN_MIN 0
+#define IN_MAX UCHAR_MAX
+#define SHIFT 8
+#include "mixeng_template.h"
+#undef IN_MAX
+#undef IN_MIN
+#undef BSIZE
+#undef ITYPE
+#undef SHIFT
+
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+
+/* Signed 16 bit */
+#define BSIZE 16
+#define ITYPE int
+#define IN_MIN SHRT_MIN
+#define IN_MAX SHRT_MAX
+#define SIGNED
+#define SHIFT 16
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap16 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#undef SIGNED
+#undef IN_MAX
+#undef IN_MIN
+#undef BSIZE
+#undef ITYPE
+#undef SHIFT
+
+/* Unsigned 16 bit */
+#define BSIZE 16
+#define ITYPE uint
+#define IN_MIN 0
+#define IN_MAX USHRT_MAX
+#define SHIFT 16
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap16 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#undef IN_MAX
+#undef IN_MIN
+#undef BSIZE
+#undef ITYPE
+#undef SHIFT
+
+/* Signed 32 bit */
+#define BSIZE 32
+#define ITYPE int
+#define IN_MIN INT32_MIN
+#define IN_MAX INT32_MAX
+#define SIGNED
+#define SHIFT 32
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap32 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#undef SIGNED
+#undef IN_MAX
+#undef IN_MIN
+#undef BSIZE
+#undef ITYPE
+#undef SHIFT
+
+/* Unsigned 32 bit */
+#define BSIZE 32
+#define ITYPE uint
+#define IN_MIN 0
+#define IN_MAX UINT32_MAX
+#define SHIFT 32
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap32 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#undef IN_MAX
+#undef IN_MIN
+#undef BSIZE
+#undef ITYPE
+#undef SHIFT
+
+t_sample *mixeng_conv[2][2][2][3] = {
+    {
+        {
+            {
+                conv_natural_uint8_t_to_mono,
+                conv_natural_uint16_t_to_mono,
+                conv_natural_uint32_t_to_mono
+            },
+            {
+                conv_natural_uint8_t_to_mono,
+                conv_swap_uint16_t_to_mono,
+                conv_swap_uint32_t_to_mono,
+            }
+        },
+        {
+            {
+                conv_natural_int8_t_to_mono,
+                conv_natural_int16_t_to_mono,
+                conv_natural_int32_t_to_mono
+            },
+            {
+                conv_natural_int8_t_to_mono,
+                conv_swap_int16_t_to_mono,
+                conv_swap_int32_t_to_mono
+            }
+        }
+    },
+    {
+        {
+            {
+                conv_natural_uint8_t_to_stereo,
+                conv_natural_uint16_t_to_stereo,
+                conv_natural_uint32_t_to_stereo
+            },
+            {
+                conv_natural_uint8_t_to_stereo,
+                conv_swap_uint16_t_to_stereo,
+                conv_swap_uint32_t_to_stereo
+            }
+        },
+        {
+            {
+                conv_natural_int8_t_to_stereo,
+                conv_natural_int16_t_to_stereo,
+                conv_natural_int32_t_to_stereo
+            },
+            {
+                conv_natural_int8_t_to_stereo,
+                conv_swap_int16_t_to_stereo,
+                conv_swap_int32_t_to_stereo,
+            }
+        }
+    }
+};
+
+f_sample *mixeng_clip[2][2][2][3] = {
+    {
+        {
+            {
+                clip_natural_uint8_t_from_mono,
+                clip_natural_uint16_t_from_mono,
+                clip_natural_uint32_t_from_mono
+            },
+            {
+                clip_natural_uint8_t_from_mono,
+                clip_swap_uint16_t_from_mono,
+                clip_swap_uint32_t_from_mono
+            }
+        },
+        {
+            {
+                clip_natural_int8_t_from_mono,
+                clip_natural_int16_t_from_mono,
+                clip_natural_int32_t_from_mono
+            },
+            {
+                clip_natural_int8_t_from_mono,
+                clip_swap_int16_t_from_mono,
+                clip_swap_int32_t_from_mono
+            }
+        }
+    },
+    {
+        {
+            {
+                clip_natural_uint8_t_from_stereo,
+                clip_natural_uint16_t_from_stereo,
+                clip_natural_uint32_t_from_stereo
+            },
+            {
+                clip_natural_uint8_t_from_stereo,
+                clip_swap_uint16_t_from_stereo,
+                clip_swap_uint32_t_from_stereo
+            }
+        },
+        {
+            {
+                clip_natural_int8_t_from_stereo,
+                clip_natural_int16_t_from_stereo,
+                clip_natural_int32_t_from_stereo
+            },
+            {
+                clip_natural_int8_t_from_stereo,
+                clip_swap_int16_t_from_stereo,
+                clip_swap_int32_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 */
+struct rate {
+    uint64_t opos;
+    uint64_t opos_inc;
+    uint32_t ipos;              /* position in the input stream (integer) */
+    struct st_sample ilast;          /* last sample in the input stream */
+};
+
+/*
+ * Prepare processing.
+ */
+void *st_rate_start (int inrate, int outrate)
+{
+    struct rate *rate = audio_calloc (AUDIO_FUNC, 1, sizeof (*rate));
+
+    if (!rate) {
+        dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
+        return NULL;
+    }
+
+    rate->opos = 0;
+
+    /* increment */
+    rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
+
+    rate->ipos = 0;
+    rate->ilast.l = 0;
+    rate->ilast.r = 0;
+    return rate;
+}
+
+#define NAME st_rate_flow_mix
+#define OP(a, b) a += b
+#include "rate_template.h"
+
+#define NAME st_rate_flow
+#define OP(a, b) a = b
+#include "rate_template.h"
+
+void st_rate_stop (void *opaque)
+{
+    g_free (opaque);
+}
+
+void mixeng_clear (struct st_sample *buf, int len)
+{
+    memset (buf, 0, len * sizeof (struct st_sample));
+}
+
+void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol)
+{
+    if (vol->mute) {
+        mixeng_clear (buf, len);
+        return;
+    }
+
+    while (len--) {
+#ifdef FLOAT_MIXENG
+        buf->l = buf->l * vol->l;
+        buf->r = buf->r * vol->r;
+#else
+        buf->l = (buf->l * vol->l) >> 32;
+        buf->r = (buf->r * vol->r) >> 32;
+#endif
+        buf += 1;
+    }
+}
diff --git a/qemu/audio/mixeng.h b/qemu/audio/mixeng.h
new file mode 100644
index 000000000..9de443b01
--- /dev/null
+++ b/qemu/audio/mixeng.h
@@ -0,0 +1,51 @@
+/*
+ * QEMU Mixing engine header
+ *
+ * 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.
+ */
+#ifndef QEMU_MIXENG_H
+#define QEMU_MIXENG_H
+
+#ifdef FLOAT_MIXENG
+typedef float mixeng_real;
+struct mixeng_volume { int mute; mixeng_real r; mixeng_real l; };
+struct st_sample { mixeng_real l; mixeng_real r; };
+#else
+struct mixeng_volume { int mute; int64_t r; int64_t l; };
+struct st_sample { int64_t l; int64_t r; };
+#endif
+
+typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
+typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
+
+extern t_sample *mixeng_conv[2][2][2][3];
+extern f_sample *mixeng_clip[2][2][2][3];
+
+void *st_rate_start (int inrate, int outrate);
+void st_rate_flow (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
+                   int *isamp, int *osamp);
+void st_rate_flow_mix (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
+                       int *isamp, int *osamp);
+void st_rate_stop (void *opaque);
+void mixeng_clear (struct st_sample *buf, int len);
+void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
+
+#endif  /* mixeng.h */
diff --git a/qemu/audio/mixeng_template.h b/qemu/audio/mixeng_template.h
new file mode 100644
index 000000000..77cc89b9e
--- /dev/null
+++ b/qemu/audio/mixeng_template.h
@@ -0,0 +1,154 @@
+/*
+ * QEMU Mixing engine
+ *
+ * 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.
+ */
+
+/*
+ * Tusen tack till Mike Nordell
+ * dec++'ified by Dscho
+ */
+
+#ifndef SIGNED
+#define HALF (IN_MAX >> 1)
+#endif
+
+#define ET glue (ENDIAN_CONVERSION, glue (glue (glue (_, ITYPE), BSIZE), _t))
+#define IN_T glue (glue (ITYPE, BSIZE), _t)
+
+#ifdef FLOAT_MIXENG
+static inline mixeng_real glue (conv_, ET) (IN_T v)
+{
+    IN_T nv = ENDIAN_CONVERT (v);
+
+#ifdef RECIPROCAL
+#ifdef SIGNED
+    return nv * (1.f / (mixeng_real) (IN_MAX - IN_MIN));
+#else
+    return (nv - HALF) * (1.f / (mixeng_real) IN_MAX);
+#endif
+#else  /* !RECIPROCAL */
+#ifdef SIGNED
+    return nv / (mixeng_real) ((mixeng_real) IN_MAX - IN_MIN);
+#else
+    return (nv - HALF) / (mixeng_real) IN_MAX;
+#endif
+#endif
+}
+
+static inline IN_T glue (clip_, ET) (mixeng_real v)
+{
+    if (v >= 0.5) {
+        return IN_MAX;
+    }
+    else if (v < -0.5) {
+        return IN_MIN;
+    }
+
+#ifdef SIGNED
+    return ENDIAN_CONVERT ((IN_T) (v * ((mixeng_real) IN_MAX - IN_MIN)));
+#else
+    return ENDIAN_CONVERT ((IN_T) ((v * IN_MAX) + HALF));
+#endif
+}
+
+#else  /* !FLOAT_MIXENG */
+
+static inline int64_t glue (conv_, ET) (IN_T v)
+{
+    IN_T nv = ENDIAN_CONVERT (v);
+#ifdef SIGNED
+    return ((int64_t) nv) << (32 - SHIFT);
+#else
+    return ((int64_t) nv - HALF) << (32 - SHIFT);
+#endif
+}
+
+static inline IN_T glue (clip_, ET) (int64_t v)
+{
+    if (v >= 0x7f000000) {
+        return IN_MAX;
+    }
+    else if (v < -2147483648LL) {
+        return IN_MIN;
+    }
+
+#ifdef SIGNED
+    return ENDIAN_CONVERT ((IN_T) (v >> (32 - SHIFT)));
+#else
+    return ENDIAN_CONVERT ((IN_T) ((v >> (32 - SHIFT)) + HALF));
+#endif
+}
+#endif
+
+static void glue (glue (conv_, ET), _to_stereo)
+    (struct st_sample *dst, const void *src, int samples)
+{
+    struct st_sample *out = dst;
+    IN_T *in = (IN_T *) src;
+
+    while (samples--) {
+        out->l = glue (conv_, ET) (*in++);
+        out->r = glue (conv_, ET) (*in++);
+        out += 1;
+    }
+}
+
+static void glue (glue (conv_, ET), _to_mono)
+    (struct st_sample *dst, const void *src, int samples)
+{
+    struct st_sample *out = dst;
+    IN_T *in = (IN_T *) src;
+
+    while (samples--) {
+        out->l = glue (conv_, ET) (in[0]);
+        out->r = out->l;
+        out += 1;
+        in += 1;
+    }
+}
+
+static void glue (glue (clip_, ET), _from_stereo)
+    (void *dst, const struct st_sample *src, int samples)
+{
+    const struct st_sample *in = src;
+    IN_T *out = (IN_T *) dst;
+    while (samples--) {
+        *out++ = glue (clip_, ET) (in->l);
+        *out++ = glue (clip_, ET) (in->r);
+        in += 1;
+    }
+}
+
+static void glue (glue (clip_, ET), _from_mono)
+    (void *dst, const struct st_sample *src, int samples)
+{
+    const struct st_sample *in = src;
+    IN_T *out = (IN_T *) dst;
+    while (samples--) {
+        *out++ = glue (clip_, ET) (in->l + in->r);
+        in += 1;
+    }
+}
+
+#undef ET
+#undef HALF
+#undef IN_T
diff --git a/qemu/audio/noaudio.c b/qemu/audio/noaudio.c
new file mode 100644
index 000000000..50db1f344
--- /dev/null
+++ b/qemu/audio/noaudio.c
@@ -0,0 +1,173 @@
+/*
+ * QEMU Timer based audio emulation
+ *
+ * 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 "qemu-common.h"
+#include "audio.h"
+#include "qemu/timer.h"
+
+#define AUDIO_CAP "noaudio"
+#include "audio_int.h"
+
+typedef struct NoVoiceOut {
+    HWVoiceOut hw;
+    int64_t old_ticks;
+} NoVoiceOut;
+
+typedef struct NoVoiceIn {
+    HWVoiceIn hw;
+    int64_t old_ticks;
+} NoVoiceIn;
+
+static int no_run_out (HWVoiceOut *hw, int live)
+{
+    NoVoiceOut *no = (NoVoiceOut *) hw;
+    int decr, samples;
+    int64_t now;
+    int64_t ticks;
+    int64_t bytes;
+
+    now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+    ticks = now - no->old_ticks;
+    bytes = muldiv64 (ticks, hw->info.bytes_per_second, get_ticks_per_sec ());
+    bytes = audio_MIN (bytes, INT_MAX);
+    samples = bytes >> hw->info.shift;
+
+    no->old_ticks = now;
+    decr = audio_MIN (live, samples);
+    hw->rpos = (hw->rpos + decr) % hw->samples;
+    return decr;
+}
+
+static int no_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
+{
+    audio_pcm_init_info (&hw->info, as);
+    hw->samples = 1024;
+    return 0;
+}
+
+static void no_fini_out (HWVoiceOut *hw)
+{
+    (void) hw;
+}
+
+static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+    audio_pcm_init_info (&hw->info, as);
+    hw->samples = 1024;
+    return 0;
+}
+
+static void no_fini_in (HWVoiceIn *hw)
+{
+    (void) hw;
+}
+
+static int no_run_in (HWVoiceIn *hw)
+{
+    NoVoiceIn *no = (NoVoiceIn *) hw;
+    int live = audio_pcm_hw_get_live_in (hw);
+    int dead = hw->samples - live;
+    int samples = 0;
+
+    if (dead) {
+        int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+        int64_t ticks = now - no->old_ticks;
+        int64_t bytes =
+            muldiv64 (ticks, hw->info.bytes_per_second, get_ticks_per_sec ());
+
+        no->old_ticks = now;
+        bytes = audio_MIN (bytes, INT_MAX);
+        samples = bytes >> hw->info.shift;
+        samples = audio_MIN (samples, dead);
+    }
+    return samples;
+}
+
+static int no_read (SWVoiceIn *sw, void *buf, int size)
+{
+    /* use custom code here instead of audio_pcm_sw_read() to avoid
+     * useless resampling/mixing */
+    int samples = size >> sw->info.shift;
+    int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+    int to_clear = audio_MIN (samples, total);
+    sw->total_hw_samples_acquired += total;
+    audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
+    return to_clear << sw->info.shift;
+}
+
+static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static void *no_audio_init (void)
+{
+    return &no_audio_init;
+}
+
+static void no_audio_fini (void *opaque)
+{
+    (void) opaque;
+}
+
+static struct audio_pcm_ops no_pcm_ops = {
+    .init_out = no_init_out,
+    .fini_out = no_fini_out,
+    .run_out  = no_run_out,
+    .write    = no_write,
+    .ctl_out  = no_ctl_out,
+
+    .init_in  = no_init_in,
+    .fini_in  = no_fini_in,
+    .run_in   = no_run_in,
+    .read     = no_read,
+    .ctl_in   = no_ctl_in
+};
+
+struct audio_driver no_audio_driver = {
+    .name           = "none",
+    .descr          = "Timer based audio emulation",
+    .options        = NULL,
+    .init           = no_audio_init,
+    .fini           = no_audio_fini,
+    .pcm_ops        = &no_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
+    .voice_size_out = sizeof (NoVoiceOut),
+    .voice_size_in  = sizeof (NoVoiceIn)
+};
diff --git a/qemu/audio/ossaudio.c b/qemu/audio/ossaudio.c
new file mode 100644
index 000000000..7dbe3332d
--- /dev/null
+++ b/qemu/audio/ossaudio.c
@@ -0,0 +1,941 @@
+/*
+ * QEMU OSS audio driver
+ *
+ * Copyright (c) 2003-2005 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 <stdlib.h>
+#include <sys/mman.h>
+#include <sys/types.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+#include "qemu-common.h"
+#include "qemu/main-loop.h"
+#include "qemu/host-utils.h"
+#include "audio.h"
+#include "trace.h"
+
+#define AUDIO_CAP "oss"
+#include "audio_int.h"
+
+#if defined OSS_GETVERSION && defined SNDCTL_DSP_POLICY
+#define USE_DSP_POLICY
+#endif
+
+typedef struct OSSConf {
+    int try_mmap;
+    int nfrags;
+    int fragsize;
+    const char *devpath_out;
+    const char *devpath_in;
+    int exclusive;
+    int policy;
+} OSSConf;
+
+typedef struct OSSVoiceOut {
+    HWVoiceOut hw;
+    void *pcm_buf;
+    int fd;
+    int wpos;
+    int nfrags;
+    int fragsize;
+    int mmapped;
+    int pending;
+    OSSConf *conf;
+} OSSVoiceOut;
+
+typedef struct OSSVoiceIn {
+    HWVoiceIn hw;
+    void *pcm_buf;
+    int fd;
+    int nfrags;
+    int fragsize;
+    OSSConf *conf;
+} OSSVoiceIn;
+
+struct oss_params {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    int nfrags;
+    int fragsize;
+};
+
+static void GCC_FMT_ATTR (2, 3) oss_logerr (int err, 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", strerror (err));
+}
+
+static void GCC_FMT_ATTR (3, 4) oss_logerr2 (
+    int err,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
+}
+
+static void oss_anal_close (int *fdp)
+{
+    int err;
+
+    qemu_set_fd_handler (*fdp, NULL, NULL, NULL);
+    err = close (*fdp);
+    if (err) {
+        oss_logerr (errno, "Failed to close file(fd=%d)\n", *fdp);
+    }
+    *fdp = -1;
+}
+
+static void oss_helper_poll_out (void *opaque)
+{
+    (void) opaque;
+    audio_run ("oss_poll_out");
+}
+
+static void oss_helper_poll_in (void *opaque)
+{
+    (void) opaque;
+    audio_run ("oss_poll_in");
+}
+
+static void oss_poll_out (HWVoiceOut *hw)
+{
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+
+    qemu_set_fd_handler (oss->fd, NULL, oss_helper_poll_out, NULL);
+}
+
+static void oss_poll_in (HWVoiceIn *hw)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+
+    qemu_set_fd_handler (oss->fd, oss_helper_poll_in, NULL, NULL);
+}
+
+static int oss_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int aud_to_ossfmt (audfmt_e fmt, int endianness)
+{
+    switch (fmt) {
+    case AUD_FMT_S8:
+        return AFMT_S8;
+
+    case AUD_FMT_U8:
+        return AFMT_U8;
+
+    case AUD_FMT_S16:
+        if (endianness) {
+            return AFMT_S16_BE;
+        }
+        else {
+            return AFMT_S16_LE;
+        }
+
+    case AUD_FMT_U16:
+        if (endianness) {
+            return AFMT_U16_BE;
+        }
+        else {
+            return AFMT_U16_LE;
+        }
+
+    default:
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+        abort ();
+#endif
+        return AFMT_U8;
+    }
+}
+
+static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
+{
+    switch (ossfmt) {
+    case AFMT_S8:
+        *endianness = 0;
+        *fmt = AUD_FMT_S8;
+        break;
+
+    case AFMT_U8:
+        *endianness = 0;
+        *fmt = AUD_FMT_U8;
+        break;
+
+    case AFMT_S16_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AFMT_U16_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case AFMT_S16_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AFMT_U16_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    default:
+        dolog ("Unrecognized audio format %d\n", ossfmt);
+        return -1;
+    }
+
+    return 0;
+}
+
+#if defined DEBUG_MISMATCHES || defined DEBUG
+static void oss_dump_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
+
+#ifdef USE_DSP_POLICY
+static int oss_get_version (int fd, int *version, const char *typ)
+{
+    if (ioctl (fd, OSS_GETVERSION, &version)) {
+#if defined(__FreeBSD__) || defined(__FreeBSD_kernel__)
+        /*
+         * Looks like atm (20100109) FreeBSD knows OSS_GETVERSION
+         * since 7.x, but currently only on the mixer device (or in
+         * the Linuxolator), and in the native version that part of
+         * the code is in fact never reached so the ioctl fails anyway.
+         * Until this is fixed, just check the errno and if its what
+         * FreeBSD's sound drivers return atm assume they are new enough.
+         */
+        if (errno == EINVAL) {
+            *version = 0x040000;
+            return 0;
+        }
+#endif
+        oss_logerr2 (errno, typ, "Failed to get OSS version\n");
+        return -1;
+    }
+    return 0;
+}
+#endif
+
+static int oss_open (int in, struct oss_params *req,
+                     struct oss_params *obt, int *pfd, OSSConf* conf)
+{
+    int fd;
+    int oflags = conf->exclusive ? O_EXCL : 0;
+    audio_buf_info abinfo;
+    int fmt, freq, nchannels;
+    int setfragment = 1;
+    const char *dspname = in ? conf->devpath_in : conf->devpath_out;
+    const char *typ = in ? "ADC" : "DAC";
+
+    /* Kludge needed to have working mmap on Linux */
+    oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+
+    fd = open (dspname, oflags | O_NONBLOCK);
+    if (-1 == fd) {
+        oss_logerr2 (errno, typ, "Failed to open `%s'\n", dspname);
+        return -1;
+    }
+
+    freq = req->freq;
+    nchannels = req->nchannels;
+    fmt = req->fmt;
+
+    if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
+        oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) {
+        oss_logerr2 (errno, typ, "Failed to set number of channels %d\n",
+                     req->nchannels);
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) {
+        oss_logerr2 (errno, typ, "Failed to set frequency %d\n", req->freq);
+        goto err;
+    }
+
+    if (ioctl (fd, SNDCTL_DSP_NONBLOCK, NULL)) {
+        oss_logerr2 (errno, typ, "Failed to set non-blocking mode\n");
+        goto err;
+    }
+
+#ifdef USE_DSP_POLICY
+    if (conf->policy >= 0) {
+        int version;
+
+        if (!oss_get_version (fd, &version, typ)) {
+            trace_oss_version(version);
+
+            if (version >= 0x040000) {
+                int policy = conf->policy;
+                if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
+                    oss_logerr2 (errno, typ,
+                                 "Failed to set timing policy to %d\n",
+                                 conf->policy);
+                    goto err;
+                }
+                setfragment = 0;
+            }
+        }
+    }
+#endif
+
+    if (setfragment) {
+        int mmmmssss = (req->nfrags << 16) | ctz32 (req->fragsize);
+        if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) {
+            oss_logerr2 (errno, typ, "Failed to set buffer length (%d, %d)\n",
+                         req->nfrags, req->fragsize);
+            goto err;
+        }
+    }
+
+    if (ioctl (fd, in ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo)) {
+        oss_logerr2 (errno, typ, "Failed to get buffer length\n");
+        goto err;
+    }
+
+    if (!abinfo.fragstotal || !abinfo.fragsize) {
+        AUD_log (AUDIO_CAP, "Returned bogus buffer information(%d, %d) for %s\n",
+                 abinfo.fragstotal, abinfo.fragsize, typ);
+        goto err;
+    }
+
+    obt->fmt = fmt;
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->nfrags = abinfo.fragstotal;
+    obt->fragsize = abinfo.fragsize;
+    *pfd = fd;
+
+#ifdef DEBUG_MISMATCHES
+    if ((req->fmt != obt->fmt) ||
+        (req->nchannels != obt->nchannels) ||
+        (req->freq != obt->freq) ||
+        (req->fragsize != obt->fragsize) ||
+        (req->nfrags != obt->nfrags)) {
+        dolog ("Audio parameters mismatch\n");
+        oss_dump_info (req, obt);
+    }
+#endif
+
+#ifdef DEBUG
+    oss_dump_info (req, obt);
+#endif
+    return 0;
+
+ err:
+    oss_anal_close (&fd);
+    return -1;
+}
+
+static void oss_write_pending (OSSVoiceOut *oss)
+{
+    HWVoiceOut *hw = &oss->hw;
+
+    if (oss->mmapped) {
+        return;
+    }
+
+    while (oss->pending) {
+        int samples_written;
+        ssize_t bytes_written;
+        int samples_till_end = hw->samples - oss->wpos;
+        int samples_to_write = audio_MIN (oss->pending, samples_till_end);
+        int bytes_to_write = samples_to_write << hw->info.shift;
+        void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift);
+
+        bytes_written = write (oss->fd, pcm, bytes_to_write);
+        if (bytes_written < 0) {
+            if (errno != EAGAIN) {
+                oss_logerr (errno, "failed to write %d bytes\n",
+                            bytes_to_write);
+            }
+            break;
+        }
+
+        if (bytes_written & hw->info.align) {
+            dolog ("misaligned write asked for %d, but got %zd\n",
+                   bytes_to_write, bytes_written);
+            return;
+        }
+
+        samples_written = bytes_written >> hw->info.shift;
+        oss->pending -= samples_written;
+        oss->wpos = (oss->wpos + samples_written) % hw->samples;
+        if (bytes_written - bytes_to_write) {
+            break;
+        }
+    }
+}
+
+static int oss_run_out (HWVoiceOut *hw, int live)
+{
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+    int err, decr;
+    struct audio_buf_info abinfo;
+    struct count_info cntinfo;
+    int bufsize;
+
+    bufsize = hw->samples << hw->info.shift;
+
+    if (oss->mmapped) {
+        int bytes, pos;
+
+        err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
+        if (err < 0) {
+            oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
+            return 0;
+        }
+
+        pos = hw->rpos << hw->info.shift;
+        bytes = audio_ring_dist (cntinfo.ptr, pos, bufsize);
+        decr = audio_MIN (bytes >> hw->info.shift, live);
+    }
+    else {
+        err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
+        if (err < 0) {
+            oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
+            return 0;
+        }
+
+        if (abinfo.bytes > bufsize) {
+            trace_oss_invalid_available_size(abinfo.bytes, bufsize);
+            abinfo.bytes = bufsize;
+        }
+
+        if (abinfo.bytes < 0) {
+            trace_oss_invalid_available_size(abinfo.bytes, bufsize);
+            return 0;
+        }
+
+        decr = audio_MIN (abinfo.bytes >> hw->info.shift, live);
+        if (!decr) {
+            return 0;
+        }
+    }
+
+    decr = audio_pcm_hw_clip_out (hw, oss->pcm_buf, decr, oss->pending);
+    oss->pending += decr;
+    oss_write_pending (oss);
+
+    return decr;
+}
+
+static void oss_fini_out (HWVoiceOut *hw)
+{
+    int err;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+
+    ldebug ("oss_fini\n");
+    oss_anal_close (&oss->fd);
+
+    if (oss->pcm_buf) {
+        if (oss->mmapped) {
+            err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
+            if (err) {
+                oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
+                            oss->pcm_buf, hw->samples << hw->info.shift);
+            }
+        }
+        else {
+            g_free (oss->pcm_buf);
+        }
+        oss->pcm_buf = NULL;
+    }
+}
+
+static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
+                        void *drv_opaque)
+{
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+    struct oss_params req, obt;
+    int endianness;
+    int err;
+    int fd;
+    audfmt_e effective_fmt;
+    struct audsettings obt_as;
+    OSSConf *conf = drv_opaque;
+
+    oss->fd = -1;
+
+    req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
+    req.freq = as->freq;
+    req.nchannels = as->nchannels;
+    req.fragsize = conf->fragsize;
+    req.nfrags = conf->nfrags;
+
+    if (oss_open (0, &req, &obt, &fd, conf)) {
+        return -1;
+    }
+
+    err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    obt_as.freq = obt.freq;
+    obt_as.nchannels = obt.nchannels;
+    obt_as.fmt = effective_fmt;
+    obt_as.endianness = endianness;
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    oss->nfrags = obt.nfrags;
+    oss->fragsize = obt.fragsize;
+
+    if (obt.nfrags * obt.fragsize & hw->info.align) {
+        dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
+               obt.nfrags * obt.fragsize, hw->info.align + 1);
+    }
+
+    hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+
+    oss->mmapped = 0;
+    if (conf->try_mmap) {
+        oss->pcm_buf = mmap (
+            NULL,
+            hw->samples << hw->info.shift,
+            PROT_READ | PROT_WRITE,
+            MAP_SHARED,
+            fd,
+            0
+            );
+        if (oss->pcm_buf == MAP_FAILED) {
+            oss_logerr (errno, "Failed to map %d bytes of DAC\n",
+                        hw->samples << hw->info.shift);
+        }
+        else {
+            int err;
+            int trig = 0;
+            if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
+            }
+            else {
+                trig = PCM_ENABLE_OUTPUT;
+                if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                    oss_logerr (
+                        errno,
+                        "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                        );
+                }
+                else {
+                    oss->mmapped = 1;
+                }
+            }
+
+            if (!oss->mmapped) {
+                err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
+                if (err) {
+                    oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
+                                oss->pcm_buf, hw->samples << hw->info.shift);
+                }
+            }
+        }
+    }
+
+    if (!oss->mmapped) {
+        oss->pcm_buf = audio_calloc (
+            AUDIO_FUNC,
+            hw->samples,
+            1 << hw->info.shift
+            );
+        if (!oss->pcm_buf) {
+            dolog (
+                "Could not allocate DAC buffer (%d samples, each %d bytes)\n",
+                hw->samples,
+                1 << hw->info.shift
+                );
+            oss_anal_close (&fd);
+            return -1;
+        }
+    }
+
+    oss->fd = fd;
+    oss->conf = conf;
+    return 0;
+}
+
+static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    int trig;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        {
+            va_list ap;
+            int poll_mode;
+
+            va_start (ap, cmd);
+            poll_mode = va_arg (ap, int);
+            va_end (ap);
+
+            ldebug ("enabling voice\n");
+            if (poll_mode) {
+                oss_poll_out (hw);
+                poll_mode = 0;
+            }
+            hw->poll_mode = poll_mode;
+
+            if (!oss->mmapped) {
+                return 0;
+            }
+
+            audio_pcm_info_clear_buf (&hw->info, oss->pcm_buf, hw->samples);
+            trig = PCM_ENABLE_OUTPUT;
+            if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                oss_logerr (
+                    errno,
+                    "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                    );
+                return -1;
+            }
+        }
+        break;
+
+    case VOICE_DISABLE:
+        if (hw->poll_mode) {
+            qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
+            hw->poll_mode = 0;
+        }
+
+        if (!oss->mmapped) {
+            return 0;
+        }
+
+        ldebug ("disabling voice\n");
+        trig = 0;
+        if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+            oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
+            return -1;
+        }
+        break;
+    }
+    return 0;
+}
+
+static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+    struct oss_params req, obt;
+    int endianness;
+    int err;
+    int fd;
+    audfmt_e effective_fmt;
+    struct audsettings obt_as;
+    OSSConf *conf = drv_opaque;
+
+    oss->fd = -1;
+
+    req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
+    req.freq = as->freq;
+    req.nchannels = as->nchannels;
+    req.fragsize = conf->fragsize;
+    req.nfrags = conf->nfrags;
+    if (oss_open (1, &req, &obt, &fd, conf)) {
+        return -1;
+    }
+
+    err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    obt_as.freq = obt.freq;
+    obt_as.nchannels = obt.nchannels;
+    obt_as.fmt = effective_fmt;
+    obt_as.endianness = endianness;
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    oss->nfrags = obt.nfrags;
+    oss->fragsize = obt.fragsize;
+
+    if (obt.nfrags * obt.fragsize & hw->info.align) {
+        dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
+               obt.nfrags * obt.fragsize, hw->info.align + 1);
+    }
+
+    hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+    oss->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    if (!oss->pcm_buf) {
+        dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
+               hw->samples, 1 << hw->info.shift);
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    oss->fd = fd;
+    oss->conf = conf;
+    return 0;
+}
+
+static void oss_fini_in (HWVoiceIn *hw)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+
+    oss_anal_close (&oss->fd);
+
+    g_free(oss->pcm_buf);
+    oss->pcm_buf = NULL;
+}
+
+static int oss_run_in (HWVoiceIn *hw)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+    int hwshift = hw->info.shift;
+    int i;
+    int live = audio_pcm_hw_get_live_in (hw);
+    int dead = hw->samples - live;
+    size_t read_samples = 0;
+    struct {
+        int add;
+        int len;
+    } bufs[2] = {
+        { .add = hw->wpos, .len = 0 },
+        { .add = 0,        .len = 0 }
+    };
+
+    if (!dead) {
+        return 0;
+    }
+
+    if (hw->wpos + dead > hw->samples) {
+        bufs[0].len = (hw->samples - hw->wpos) << hwshift;
+        bufs[1].len = (dead - (hw->samples - hw->wpos)) << hwshift;
+    }
+    else {
+        bufs[0].len = dead << hwshift;
+    }
+
+    for (i = 0; i < 2; ++i) {
+        ssize_t nread;
+
+        if (bufs[i].len) {
+            void *p = advance (oss->pcm_buf, bufs[i].add << hwshift);
+            nread = read (oss->fd, p, bufs[i].len);
+
+            if (nread > 0) {
+                if (nread & hw->info.align) {
+                    dolog ("warning: Misaligned read %zd (requested %d), "
+                           "alignment %d\n", nread, bufs[i].add << hwshift,
+                           hw->info.align + 1);
+                }
+                read_samples += nread >> hwshift;
+                hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift);
+            }
+
+            if (bufs[i].len - nread) {
+                if (nread == -1) {
+                    switch (errno) {
+                    case EINTR:
+                    case EAGAIN:
+                        break;
+                    default:
+                        oss_logerr (
+                            errno,
+                            "Failed to read %d bytes of audio (to %p)\n",
+                            bufs[i].len, p
+                            );
+                        break;
+                    }
+                }
+                break;
+            }
+        }
+    }
+
+    hw->wpos = (hw->wpos + read_samples) % hw->samples;
+    return read_samples;
+}
+
+static int oss_read (SWVoiceIn *sw, void *buf, int size)
+{
+    return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        {
+            va_list ap;
+            int poll_mode;
+
+            va_start (ap, cmd);
+            poll_mode = va_arg (ap, int);
+            va_end (ap);
+
+            if (poll_mode) {
+                oss_poll_in (hw);
+                poll_mode = 0;
+            }
+            hw->poll_mode = poll_mode;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        if (hw->poll_mode) {
+            hw->poll_mode = 0;
+            qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
+        }
+        break;
+    }
+    return 0;
+}
+
+static OSSConf glob_conf = {
+    .try_mmap = 0,
+    .nfrags = 4,
+    .fragsize = 4096,
+    .devpath_out = "/dev/dsp",
+    .devpath_in = "/dev/dsp",
+    .exclusive = 0,
+    .policy = 5
+};
+
+static void *oss_audio_init (void)
+{
+    OSSConf *conf = g_malloc(sizeof(OSSConf));
+    *conf = glob_conf;
+
+    if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
+        access(conf->devpath_out, R_OK | W_OK) < 0) {
+        g_free(conf);
+        return NULL;
+    }
+    return conf;
+}
+
+static void oss_audio_fini (void *opaque)
+{
+    g_free(opaque);
+}
+
+static struct audio_option oss_options[] = {
+    {
+        .name  = "FRAGSIZE",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.fragsize,
+        .descr = "Fragment size in bytes"
+    },
+    {
+        .name  = "NFRAGS",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.nfrags,
+        .descr = "Number of fragments"
+    },
+    {
+        .name  = "MMAP",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &glob_conf.try_mmap,
+        .descr = "Try using memory mapped access"
+    },
+    {
+        .name  = "DAC_DEV",
+        .tag   = AUD_OPT_STR,
+        .valp  = &glob_conf.devpath_out,
+        .descr = "Path to DAC device"
+    },
+    {
+        .name  = "ADC_DEV",
+        .tag   = AUD_OPT_STR,
+        .valp  = &glob_conf.devpath_in,
+        .descr = "Path to ADC device"
+    },
+    {
+        .name  = "EXCLUSIVE",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &glob_conf.exclusive,
+        .descr = "Open device in exclusive mode (vmix wont work)"
+    },
+#ifdef USE_DSP_POLICY
+    {
+        .name  = "POLICY",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.policy,
+        .descr = "Set the timing policy of the device, -1 to use fragment mode",
+    },
+#endif
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops oss_pcm_ops = {
+    .init_out = oss_init_out,
+    .fini_out = oss_fini_out,
+    .run_out  = oss_run_out,
+    .write    = oss_write,
+    .ctl_out  = oss_ctl_out,
+
+    .init_in  = oss_init_in,
+    .fini_in  = oss_fini_in,
+    .run_in   = oss_run_in,
+    .read     = oss_read,
+    .ctl_in   = oss_ctl_in
+};
+
+struct audio_driver oss_audio_driver = {
+    .name           = "oss",
+    .descr          = "OSS http://www.opensound.com",
+    .options        = oss_options,
+    .init           = oss_audio_init,
+    .fini           = oss_audio_fini,
+    .pcm_ops        = &oss_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
+    .voice_size_out = sizeof (OSSVoiceOut),
+    .voice_size_in  = sizeof (OSSVoiceIn)
+};
diff --git a/qemu/audio/paaudio.c b/qemu/audio/paaudio.c
new file mode 100644
index 000000000..fea607166
--- /dev/null
+++ b/qemu/audio/paaudio.c
@@ -0,0 +1,953 @@
+/* public domain */
+#include "qemu-common.h"
+#include "audio.h"
+
+#include <pulse/pulseaudio.h>
+
+#define AUDIO_CAP "pulseaudio"
+#include "audio_int.h"
+#include "audio_pt_int.h"
+
+typedef struct {
+    int samples;
+    char *server;
+    char *sink;
+    char *source;
+} PAConf;
+
+typedef struct {
+    PAConf conf;
+    pa_threaded_mainloop *mainloop;
+    pa_context *context;
+} paaudio;
+
+typedef struct {
+    HWVoiceOut hw;
+    int done;
+    int live;
+    int decr;
+    int rpos;
+    pa_stream *stream;
+    void *pcm_buf;
+    struct audio_pt pt;
+    paaudio *g;
+} PAVoiceOut;
+
+typedef struct {
+    HWVoiceIn hw;
+    int done;
+    int dead;
+    int incr;
+    int wpos;
+    pa_stream *stream;
+    void *pcm_buf;
+    struct audio_pt pt;
+    const void *read_data;
+    size_t read_index, read_length;
+    paaudio *g;
+} PAVoiceIn;
+
+static void qpa_audio_fini(void *opaque);
+
+static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, 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", pa_strerror (err));
+}
+
+#ifndef PA_CONTEXT_IS_GOOD
+static inline int PA_CONTEXT_IS_GOOD(pa_context_state_t x)
+{
+    return
+        x == PA_CONTEXT_CONNECTING ||
+        x == PA_CONTEXT_AUTHORIZING ||
+        x == PA_CONTEXT_SETTING_NAME ||
+        x == PA_CONTEXT_READY;
+}
+#endif
+
+#ifndef PA_STREAM_IS_GOOD
+static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
+{
+    return
+        x == PA_STREAM_CREATING ||
+        x == PA_STREAM_READY;
+}
+#endif
+
+#define CHECK_SUCCESS_GOTO(c, rerror, expression, label)        \
+    do {                                                        \
+        if (!(expression)) {                                    \
+            if (rerror) {                                       \
+                *(rerror) = pa_context_errno ((c)->context);    \
+            }                                                   \
+            goto label;                                         \
+        }                                                       \
+    } while (0);
+
+#define CHECK_DEAD_GOTO(c, stream, rerror, label)                       \
+    do {                                                                \
+        if (!(c)->context || !PA_CONTEXT_IS_GOOD (pa_context_get_state((c)->context)) || \
+            !(stream) || !PA_STREAM_IS_GOOD (pa_stream_get_state ((stream)))) { \
+            if (((c)->context && pa_context_get_state ((c)->context) == PA_CONTEXT_FAILED) || \
+                ((stream) && pa_stream_get_state ((stream)) == PA_STREAM_FAILED)) { \
+                if (rerror) {                                           \
+                    *(rerror) = pa_context_errno ((c)->context);        \
+                }                                                       \
+            } else {                                                    \
+                if (rerror) {                                           \
+                    *(rerror) = PA_ERR_BADSTATE;                        \
+                }                                                       \
+            }                                                           \
+            goto label;                                                 \
+        }                                                               \
+    } while (0);
+
+static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
+{
+    paaudio *g = p->g;
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+
+    while (length > 0) {
+        size_t l;
+
+        while (!p->read_data) {
+            int r;
+
+            r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
+            CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+
+            if (!p->read_data) {
+                pa_threaded_mainloop_wait (g->mainloop);
+                CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+            } else {
+                p->read_index = 0;
+            }
+        }
+
+        l = p->read_length < length ? p->read_length : length;
+        memcpy (data, (const uint8_t *) p->read_data+p->read_index, l);
+
+        data = (uint8_t *) data + l;
+        length -= l;
+
+        p->read_index += l;
+        p->read_length -= l;
+
+        if (!p->read_length) {
+            int r;
+
+            r = pa_stream_drop (p->stream);
+            p->read_data = NULL;
+            p->read_length = 0;
+            p->read_index = 0;
+
+            CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+        }
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return 0;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return -1;
+}
+
+static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
+{
+    paaudio *g = p->g;
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+
+    while (length > 0) {
+        size_t l;
+        int r;
+
+        while (!(l = pa_stream_writable_size (p->stream))) {
+            pa_threaded_mainloop_wait (g->mainloop);
+            CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+        }
+
+        CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
+
+        if (l > length) {
+            l = length;
+        }
+
+        r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
+        CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
+
+        data = (const uint8_t *) data + l;
+        length -= l;
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return 0;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return -1;
+}
+
+static void *qpa_thread_out (void *arg)
+{
+    PAVoiceOut *pa = arg;
+    HWVoiceOut *hw = &pa->hw;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return NULL;
+    }
+
+    for (;;) {
+        int decr, to_mix, rpos;
+
+        for (;;) {
+            if (pa->done) {
+                goto exit;
+            }
+
+            if (pa->live > 0) {
+                break;
+            }
+
+            if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
+                goto exit;
+            }
+        }
+
+        decr = to_mix = audio_MIN (pa->live, pa->g->conf.samples >> 2);
+        rpos = pa->rpos;
+
+        if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        while (to_mix) {
+            int error;
+            int chunk = audio_MIN (to_mix, hw->samples - rpos);
+            struct st_sample *src = hw->mix_buf + rpos;
+
+            hw->clip (pa->pcm_buf, src, chunk);
+
+            if (qpa_simple_write (pa, pa->pcm_buf,
+                                  chunk << hw->info.shift, &error) < 0) {
+                qpa_logerr (error, "pa_simple_write failed\n");
+                return NULL;
+            }
+
+            rpos = (rpos + chunk) % hw->samples;
+            to_mix -= chunk;
+        }
+
+        if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        pa->rpos = rpos;
+        pa->live -= decr;
+        pa->decr += decr;
+    }
+
+ exit:
+    audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    return NULL;
+}
+
+static int qpa_run_out (HWVoiceOut *hw, int live)
+{
+    int decr;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return 0;
+    }
+
+    decr = audio_MIN (live, pa->decr);
+    pa->decr -= decr;
+    pa->live = live - decr;
+    hw->rpos = pa->rpos;
+    if (pa->live > 0) {
+        audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    }
+    else {
+        audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    }
+    return decr;
+}
+
+static int qpa_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+/* capture */
+static void *qpa_thread_in (void *arg)
+{
+    PAVoiceIn *pa = arg;
+    HWVoiceIn *hw = &pa->hw;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return NULL;
+    }
+
+    for (;;) {
+        int incr, to_grab, wpos;
+
+        for (;;) {
+            if (pa->done) {
+                goto exit;
+            }
+
+            if (pa->dead > 0) {
+                break;
+            }
+
+            if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
+                goto exit;
+            }
+        }
+
+        incr = to_grab = audio_MIN (pa->dead, pa->g->conf.samples >> 2);
+        wpos = pa->wpos;
+
+        if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        while (to_grab) {
+            int error;
+            int chunk = audio_MIN (to_grab, hw->samples - wpos);
+            void *buf = advance (pa->pcm_buf, wpos);
+
+            if (qpa_simple_read (pa, buf,
+                                 chunk << hw->info.shift, &error) < 0) {
+                qpa_logerr (error, "pa_simple_read failed\n");
+                return NULL;
+            }
+
+            hw->conv (hw->conv_buf + wpos, buf, chunk);
+            wpos = (wpos + chunk) % hw->samples;
+            to_grab -= chunk;
+        }
+
+        if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        pa->wpos = wpos;
+        pa->dead -= incr;
+        pa->incr += incr;
+    }
+
+ exit:
+    audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    return NULL;
+}
+
+static int qpa_run_in (HWVoiceIn *hw)
+{
+    int live, incr, dead;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_in (hw);
+    dead = hw->samples - live;
+    incr = audio_MIN (dead, pa->incr);
+    pa->incr -= incr;
+    pa->dead = dead - incr;
+    hw->wpos = pa->wpos;
+    if (pa->dead > 0) {
+        audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    }
+    else {
+        audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    }
+    return incr;
+}
+
+static int qpa_read (SWVoiceIn *sw, void *buf, int len)
+{
+    return audio_pcm_sw_read (sw, buf, len);
+}
+
+static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
+{
+    int format;
+
+    switch (afmt) {
+    case AUD_FMT_S8:
+    case AUD_FMT_U8:
+        format = PA_SAMPLE_U8;
+        break;
+    case AUD_FMT_S16:
+    case AUD_FMT_U16:
+        format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
+        break;
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
+        break;
+    default:
+        dolog ("Internal logic error: Bad audio format %d\n", afmt);
+        format = PA_SAMPLE_U8;
+        break;
+    }
+    return format;
+}
+
+static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+{
+    switch (fmt) {
+    case PA_SAMPLE_U8:
+        return AUD_FMT_U8;
+    case PA_SAMPLE_S16BE:
+        *endianness = 1;
+        return AUD_FMT_S16;
+    case PA_SAMPLE_S16LE:
+        *endianness = 0;
+        return AUD_FMT_S16;
+    case PA_SAMPLE_S32BE:
+        *endianness = 1;
+        return AUD_FMT_S32;
+    case PA_SAMPLE_S32LE:
+        *endianness = 0;
+        return AUD_FMT_S32;
+    default:
+        dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
+        return AUD_FMT_U8;
+    }
+}
+
+static void context_state_cb (pa_context *c, void *userdata)
+{
+    paaudio *g = userdata;
+
+    switch (pa_context_get_state(c)) {
+    case PA_CONTEXT_READY:
+    case PA_CONTEXT_TERMINATED:
+    case PA_CONTEXT_FAILED:
+        pa_threaded_mainloop_signal (g->mainloop, 0);
+        break;
+
+    case PA_CONTEXT_UNCONNECTED:
+    case PA_CONTEXT_CONNECTING:
+    case PA_CONTEXT_AUTHORIZING:
+    case PA_CONTEXT_SETTING_NAME:
+        break;
+    }
+}
+
+static void stream_state_cb (pa_stream *s, void * userdata)
+{
+    paaudio *g = userdata;
+
+    switch (pa_stream_get_state (s)) {
+
+    case PA_STREAM_READY:
+    case PA_STREAM_FAILED:
+    case PA_STREAM_TERMINATED:
+        pa_threaded_mainloop_signal (g->mainloop, 0);
+        break;
+
+    case PA_STREAM_UNCONNECTED:
+    case PA_STREAM_CREATING:
+        break;
+    }
+}
+
+static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
+{
+    paaudio *g = userdata;
+
+    pa_threaded_mainloop_signal (g->mainloop, 0);
+}
+
+static pa_stream *qpa_simple_new (
+        paaudio *g,
+        const char *name,
+        pa_stream_direction_t dir,
+        const char *dev,
+        const pa_sample_spec *ss,
+        const pa_channel_map *map,
+        const pa_buffer_attr *attr,
+        int *rerror)
+{
+    int r;
+    pa_stream *stream;
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    stream = pa_stream_new (g->context, name, ss, map);
+    if (!stream) {
+        goto fail;
+    }
+
+    pa_stream_set_state_callback (stream, stream_state_cb, g);
+    pa_stream_set_read_callback (stream, stream_request_cb, g);
+    pa_stream_set_write_callback (stream, stream_request_cb, g);
+
+    if (dir == PA_STREAM_PLAYBACK) {
+        r = pa_stream_connect_playback (stream, dev, attr,
+                                        PA_STREAM_INTERPOLATE_TIMING
+#ifdef PA_STREAM_ADJUST_LATENCY
+                                        |PA_STREAM_ADJUST_LATENCY
+#endif
+                                        |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
+    } else {
+        r = pa_stream_connect_record (stream, dev, attr,
+                                      PA_STREAM_INTERPOLATE_TIMING
+#ifdef PA_STREAM_ADJUST_LATENCY
+                                      |PA_STREAM_ADJUST_LATENCY
+#endif
+                                      |PA_STREAM_AUTO_TIMING_UPDATE);
+    }
+
+    if (r < 0) {
+      goto fail;
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+
+    return stream;
+
+fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+
+    if (stream) {
+        pa_stream_unref (stream);
+    }
+
+    *rerror = pa_context_errno (g->context);
+
+    return NULL;
+}
+
+static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
+                        void *drv_opaque)
+{
+    int error;
+    pa_sample_spec ss;
+    pa_buffer_attr ba;
+    struct audsettings obt_as = *as;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+    paaudio *g = pa->g = drv_opaque;
+
+    ss.format = audfmt_to_pa (as->fmt, as->endianness);
+    ss.channels = as->nchannels;
+    ss.rate = as->freq;
+
+    /*
+     * qemu audio tick runs at 100 Hz (by default), so processing
+     * data chunks worth 10 ms of sound should be a good fit.
+     */
+    ba.tlength = pa_usec_to_bytes (10 * 1000, &ss);
+    ba.minreq = pa_usec_to_bytes (5 * 1000, &ss);
+    ba.maxlength = -1;
+    ba.prebuf = -1;
+
+    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
+
+    pa->stream = qpa_simple_new (
+        g,
+        "qemu",
+        PA_STREAM_PLAYBACK,
+        g->conf.sink,
+        &ss,
+        NULL,                   /* channel map */
+        &ba,                    /* buffering attributes */
+        &error
+        );
+    if (!pa->stream) {
+        qpa_logerr (error, "pa_simple_new for playback failed\n");
+        goto fail1;
+    }
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    hw->samples = g->conf.samples;
+    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    pa->rpos = hw->rpos;
+    if (!pa->pcm_buf) {
+        dolog ("Could not allocate buffer (%d bytes)\n",
+               hw->samples << hw->info.shift);
+        goto fail2;
+    }
+
+    if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) {
+        goto fail3;
+    }
+
+    return 0;
+
+ fail3:
+    g_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+ fail2:
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
+    }
+ fail1:
+    return -1;
+}
+
+static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+    int error;
+    pa_sample_spec ss;
+    struct audsettings obt_as = *as;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+    paaudio *g = pa->g = drv_opaque;
+
+    ss.format = audfmt_to_pa (as->fmt, as->endianness);
+    ss.channels = as->nchannels;
+    ss.rate = as->freq;
+
+    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
+
+    pa->stream = qpa_simple_new (
+        g,
+        "qemu",
+        PA_STREAM_RECORD,
+        g->conf.source,
+        &ss,
+        NULL,                   /* channel map */
+        NULL,                   /* buffering attributes */
+        &error
+        );
+    if (!pa->stream) {
+        qpa_logerr (error, "pa_simple_new for capture failed\n");
+        goto fail1;
+    }
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    hw->samples = g->conf.samples;
+    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    pa->wpos = hw->wpos;
+    if (!pa->pcm_buf) {
+        dolog ("Could not allocate buffer (%d bytes)\n",
+               hw->samples << hw->info.shift);
+        goto fail2;
+    }
+
+    if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
+        goto fail3;
+    }
+
+    return 0;
+
+ fail3:
+    g_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+ fail2:
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
+    }
+ fail1:
+    return -1;
+}
+
+static void qpa_fini_out (HWVoiceOut *hw)
+{
+    void *ret;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+
+    audio_pt_lock (&pa->pt, AUDIO_FUNC);
+    pa->done = 1;
+    audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
+
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
+    }
+
+    audio_pt_fini (&pa->pt, AUDIO_FUNC);
+    g_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+}
+
+static void qpa_fini_in (HWVoiceIn *hw)
+{
+    void *ret;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+
+    audio_pt_lock (&pa->pt, AUDIO_FUNC);
+    pa->done = 1;
+    audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
+
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
+    }
+
+    audio_pt_fini (&pa->pt, AUDIO_FUNC);
+    g_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+}
+
+static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+    pa_operation *op;
+    pa_cvolume v;
+    paaudio *g = pa->g;
+
+#ifdef PA_CHECK_VERSION    /* macro is present in 0.9.16+ */
+    pa_cvolume_init (&v);  /* function is present in 0.9.13+ */
+#endif
+
+    switch (cmd) {
+    case VOICE_VOLUME:
+        {
+            SWVoiceOut *sw;
+            va_list ap;
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceOut *);
+            va_end (ap);
+
+            v.channels = 2;
+            v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
+            v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
+
+            pa_threaded_mainloop_lock (g->mainloop);
+
+            op = pa_context_set_sink_input_volume (g->context,
+                pa_stream_get_index (pa->stream),
+                &v, NULL, NULL);
+            if (!op)
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_sink_input_volume() failed\n");
+            else
+                pa_operation_unref (op);
+
+            op = pa_context_set_sink_input_mute (g->context,
+                pa_stream_get_index (pa->stream),
+               sw->vol.mute, NULL, NULL);
+            if (!op) {
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_sink_input_mute() failed\n");
+            } else {
+                pa_operation_unref (op);
+            }
+
+            pa_threaded_mainloop_unlock (g->mainloop);
+        }
+    }
+    return 0;
+}
+
+static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+    pa_operation *op;
+    pa_cvolume v;
+    paaudio *g = pa->g;
+
+#ifdef PA_CHECK_VERSION
+    pa_cvolume_init (&v);
+#endif
+
+    switch (cmd) {
+    case VOICE_VOLUME:
+        {
+            SWVoiceIn *sw;
+            va_list ap;
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceIn *);
+            va_end (ap);
+
+            v.channels = 2;
+            v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
+            v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
+
+            pa_threaded_mainloop_lock (g->mainloop);
+
+            /* FIXME: use the upcoming "set_source_output_{volume,mute}" */
+            op = pa_context_set_source_volume_by_index (g->context,
+                pa_stream_get_device_index (pa->stream),
+                &v, NULL, NULL);
+            if (!op) {
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_source_volume() failed\n");
+            } else {
+                pa_operation_unref(op);
+            }
+
+            op = pa_context_set_source_mute_by_index (g->context,
+                pa_stream_get_index (pa->stream),
+                sw->vol.mute, NULL, NULL);
+            if (!op) {
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_source_mute() failed\n");
+            } else {
+                pa_operation_unref (op);
+            }
+
+            pa_threaded_mainloop_unlock (g->mainloop);
+        }
+    }
+    return 0;
+}
+
+/* common */
+static PAConf glob_conf = {
+    .samples = 4096,
+};
+
+static void *qpa_audio_init (void)
+{
+    paaudio *g = g_malloc(sizeof(paaudio));
+    g->conf = glob_conf;
+    g->mainloop = NULL;
+    g->context = NULL;
+
+    g->mainloop = pa_threaded_mainloop_new ();
+    if (!g->mainloop) {
+        goto fail;
+    }
+
+    g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
+                                 g->conf.server);
+    if (!g->context) {
+        goto fail;
+    }
+
+    pa_context_set_state_callback (g->context, context_state_cb, g);
+
+    if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
+        qpa_logerr (pa_context_errno (g->context),
+                    "pa_context_connect() failed\n");
+        goto fail;
+    }
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    if (pa_threaded_mainloop_start (g->mainloop) < 0) {
+        goto unlock_and_fail;
+    }
+
+    for (;;) {
+        pa_context_state_t state;
+
+        state = pa_context_get_state (g->context);
+
+        if (state == PA_CONTEXT_READY) {
+            break;
+        }
+
+        if (!PA_CONTEXT_IS_GOOD (state)) {
+            qpa_logerr (pa_context_errno (g->context),
+                        "Wrong context state\n");
+            goto unlock_and_fail;
+        }
+
+        /* Wait until the context is ready */
+        pa_threaded_mainloop_wait (g->mainloop);
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+
+    return g;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+fail:
+    AUD_log (AUDIO_CAP, "Failed to initialize PA context");
+    qpa_audio_fini(g);
+    return NULL;
+}
+
+static void qpa_audio_fini (void *opaque)
+{
+    paaudio *g = opaque;
+
+    if (g->mainloop) {
+        pa_threaded_mainloop_stop (g->mainloop);
+    }
+
+    if (g->context) {
+        pa_context_disconnect (g->context);
+        pa_context_unref (g->context);
+    }
+
+    if (g->mainloop) {
+        pa_threaded_mainloop_free (g->mainloop);
+    }
+
+    g_free(g);
+}
+
+struct audio_option qpa_options[] = {
+    {
+        .name  = "SAMPLES",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.samples,
+        .descr = "buffer size in samples"
+    },
+    {
+        .name  = "SERVER",
+        .tag   = AUD_OPT_STR,
+        .valp  = &glob_conf.server,
+        .descr = "server address"
+    },
+    {
+        .name  = "SINK",
+        .tag   = AUD_OPT_STR,
+        .valp  = &glob_conf.sink,
+        .descr = "sink device name"
+    },
+    {
+        .name  = "SOURCE",
+        .tag   = AUD_OPT_STR,
+        .valp  = &glob_conf.source,
+        .descr = "source device name"
+    },
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops qpa_pcm_ops = {
+    .init_out = qpa_init_out,
+    .fini_out = qpa_fini_out,
+    .run_out  = qpa_run_out,
+    .write    = qpa_write,
+    .ctl_out  = qpa_ctl_out,
+
+    .init_in  = qpa_init_in,
+    .fini_in  = qpa_fini_in,
+    .run_in   = qpa_run_in,
+    .read     = qpa_read,
+    .ctl_in   = qpa_ctl_in
+};
+
+struct audio_driver pa_audio_driver = {
+    .name           = "pa",
+    .descr          = "http://www.pulseaudio.org/",
+    .options        = qpa_options,
+    .init           = qpa_audio_init,
+    .fini           = qpa_audio_fini,
+    .pcm_ops        = &qpa_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
+    .voice_size_out = sizeof (PAVoiceOut),
+    .voice_size_in  = sizeof (PAVoiceIn),
+    .ctl_caps       = VOICE_VOLUME_CAP
+};
diff --git a/qemu/audio/rate_template.h b/qemu/audio/rate_template.h
new file mode 100644
index 000000000..bd4b1c768
--- /dev/null
+++ b/qemu/audio/rate_template.h
@@ -0,0 +1,111 @@
+/*
+ * QEMU Mixing engine
+ *
+ * Copyright (c) 2004-2005 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.
+ */
+
+/*
+ * Processed signed long samples from ibuf to obuf.
+ * Return number of samples processed.
+ */
+void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
+           int *isamp, int *osamp)
+{
+    struct rate *rate = opaque;
+    struct st_sample *istart, *iend;
+    struct st_sample *ostart, *oend;
+    struct st_sample ilast, icur, out;
+#ifdef FLOAT_MIXENG
+    mixeng_real t;
+#else
+    int64_t t;
+#endif
+
+    ilast = rate->ilast;
+
+    istart = ibuf;
+    iend = ibuf + *isamp;
+
+    ostart = obuf;
+    oend = obuf + *osamp;
+
+    if (rate->opos_inc == (1ULL + UINT_MAX)) {
+        int i, n = *isamp > *osamp ? *osamp : *isamp;
+        for (i = 0; i < n; i++) {
+            OP (obuf[i].l, ibuf[i].l);
+            OP (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 */
+#ifdef FLOAT_MIXENG
+#ifdef RECIPROCAL
+        t = (rate->opos & UINT_MAX) * (1.f / UINT_MAX);
+#else
+        t = (rate->opos & UINT_MAX) / (mixeng_real) UINT_MAX;
+#endif
+        out.l = (ilast.l * (1.0 - t)) + icur.l * t;
+        out.r = (ilast.r * (1.0 - t)) + icur.r * t;
+#else
+        t = rate->opos & 0xffffffff;
+        out.l = (ilast.l * ((int64_t) UINT_MAX - t) + icur.l * t) >> 32;
+        out.r = (ilast.r * ((int64_t) UINT_MAX - t) + icur.r * t) >> 32;
+#endif
+
+        /* output sample & increment position */
+        OP (obuf->l, out.l);
+        OP (obuf->r, out.r);
+        obuf += 1;
+        rate->opos += rate->opos_inc;
+    }
+
+the_end:
+    *isamp = ibuf - istart;
+    *osamp = obuf - ostart;
+    rate->ilast = ilast;
+}
+
+#undef NAME
+#undef OP
diff --git a/qemu/audio/sdlaudio.c b/qemu/audio/sdlaudio.c
new file mode 100644
index 000000000..1140f2ea0
--- /dev/null
+++ b/qemu/audio/sdlaudio.c
@@ -0,0 +1,466 @@
+/*
+ * QEMU SDL audio driver
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#include <SDL.h>
+#include <SDL_thread.h>
+#include "qemu-common.h"
+#include "audio.h"
+
+#ifndef _WIN32
+#ifdef __sun__
+#define _POSIX_PTHREAD_SEMANTICS 1
+#elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__)
+#include <pthread.h>
+#endif
+#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 = {
+    .nb_samples = 1024
+};
+
+static struct SDLAudioState {
+    int exit;
+    SDL_mutex *mutex;
+    SDL_sem *sem;
+    int initialized;
+    bool driver_created;
+} 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)
+{
+    switch (fmt) {
+    case AUD_FMT_S8:
+        return AUDIO_S8;
+
+    case AUD_FMT_U8:
+        return AUDIO_U8;
+
+    case AUD_FMT_S16:
+        return AUDIO_S16LSB;
+
+    case AUD_FMT_U16:
+        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 *endianness)
+{
+    switch (sdlfmt) {
+    case AUDIO_S8:
+        *endianness = 0;
+        *fmt = AUD_FMT_S8;
+        break;
+
+    case AUDIO_U8:
+        *endianness = 0;
+        *fmt = AUD_FMT_U8;
+        break;
+
+    case AUDIO_S16LSB:
+        *endianness = 0;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AUDIO_U16LSB:
+        *endianness = 0;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case AUDIO_S16MSB:
+        *endianness = 1;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AUDIO_U16MSB:
+        *endianness = 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
+    int err;
+    sigset_t new, old;
+
+    /* Make sure potential threads created by SDL don't hog signals.  */
+    err = sigfillset (&new);
+    if (err) {
+        dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno));
+        return -1;
+    }
+    err = pthread_sigmask (SIG_BLOCK, &new, &old);
+    if (err) {
+        dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
+        return -1;
+    }
+#endif
+
+    status = SDL_OpenAudio (req, obt);
+    if (status) {
+        sdl_logerr ("SDL_OpenAudio failed\n");
+    }
+
+#ifndef _WIN32
+    err = pthread_sigmask (SIG_SETMASK, &old, NULL);
+    if (err) {
+        dolog ("sdl_open: pthread_sigmask (restore) failed: %s\n",
+               strerror (errno));
+        /* We have failed to restore original signal mask, all bets are off,
+           so exit the process */
+        exit (EXIT_FAILURE);
+    }
+#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 live)
+{
+    int decr;
+    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+    SDLAudioState *s = &glob_sdl;
+
+    if (sdl_lock (s, "sdl_run_out")) {
+        return 0;
+    }
+
+    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_run_out");
+    }
+    else {
+        sdl_unlock (s, "sdl_run_out");
+    }
+    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,
+                        void *drv_opaque)
+{
+    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+    SDLAudioState *s = &glob_sdl;
+    SDL_AudioSpec req, obt;
+    int endianness;
+    int err;
+    audfmt_e effective_fmt;
+    struct audsettings obt_as;
+
+    req.freq = as->freq;
+    req.format = aud_to_sdlfmt (as->fmt);
+    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, &endianness);
+    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 = endianness;
+
+    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 (s->driver_created) {
+        sdl_logerr("Can't create multiple sdl backends\n");
+        return NULL;
+    }
+
+    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;
+    }
+
+    s->driver_created = true;
+    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);
+    s->driver_created = false;
+}
+
+static struct audio_option sdl_options[] = {
+    {
+        .name  = "SAMPLES",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.nb_samples,
+        .descr = "Size of SDL buffer in samples"
+    },
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops sdl_pcm_ops = {
+    .init_out = sdl_init_out,
+    .fini_out = sdl_fini_out,
+    .run_out  = sdl_run_out,
+    .write    = sdl_write_out,
+    .ctl_out  = sdl_ctl_out,
+};
+
+struct audio_driver sdl_audio_driver = {
+    .name           = "sdl",
+    .descr          = "SDL http://www.libsdl.org",
+    .options        = sdl_options,
+    .init           = sdl_audio_init,
+    .fini           = sdl_audio_fini,
+    .pcm_ops        = &sdl_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = 1,
+    .max_voices_in  = 0,
+    .voice_size_out = sizeof (SDLVoiceOut),
+    .voice_size_in  = 0
+};
diff --git a/qemu/audio/spiceaudio.c b/qemu/audio/spiceaudio.c
new file mode 100644
index 000000000..42ae4a45f
--- /dev/null
+++ b/qemu/audio/spiceaudio.c
@@ -0,0 +1,411 @@
+/*
+ * Copyright (C) 2010 Red Hat, Inc.
+ *
+ * maintained by Gerd Hoffmann <kraxel@redhat.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 or
+ * (at your option) version 3 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "hw/hw.h"
+#include "qemu/error-report.h"
+#include "qemu/timer.h"
+#include "ui/qemu-spice.h"
+
+#define AUDIO_CAP "spice"
+#include "audio.h"
+#include "audio_int.h"
+
+#if SPICE_INTERFACE_PLAYBACK_MAJOR > 1 || SPICE_INTERFACE_PLAYBACK_MINOR >= 3
+#define LINE_OUT_SAMPLES (480 * 4)
+#else
+#define LINE_OUT_SAMPLES (256 * 4)
+#endif
+
+#if SPICE_INTERFACE_RECORD_MAJOR > 2 || SPICE_INTERFACE_RECORD_MINOR >= 3
+#define LINE_IN_SAMPLES (480 * 4)
+#else
+#define LINE_IN_SAMPLES (256 * 4)
+#endif
+
+typedef struct SpiceRateCtl {
+    int64_t               start_ticks;
+    int64_t               bytes_sent;
+} SpiceRateCtl;
+
+typedef struct SpiceVoiceOut {
+    HWVoiceOut            hw;
+    SpicePlaybackInstance sin;
+    SpiceRateCtl          rate;
+    int                   active;
+    uint32_t              *frame;
+    uint32_t              *fpos;
+    uint32_t              fsize;
+} SpiceVoiceOut;
+
+typedef struct SpiceVoiceIn {
+    HWVoiceIn             hw;
+    SpiceRecordInstance   sin;
+    SpiceRateCtl          rate;
+    int                   active;
+    uint32_t              samples[LINE_IN_SAMPLES];
+} SpiceVoiceIn;
+
+static const SpicePlaybackInterface playback_sif = {
+    .base.type          = SPICE_INTERFACE_PLAYBACK,
+    .base.description   = "playback",
+    .base.major_version = SPICE_INTERFACE_PLAYBACK_MAJOR,
+    .base.minor_version = SPICE_INTERFACE_PLAYBACK_MINOR,
+};
+
+static const SpiceRecordInterface record_sif = {
+    .base.type          = SPICE_INTERFACE_RECORD,
+    .base.description   = "record",
+    .base.major_version = SPICE_INTERFACE_RECORD_MAJOR,
+    .base.minor_version = SPICE_INTERFACE_RECORD_MINOR,
+};
+
+static void *spice_audio_init (void)
+{
+    if (!using_spice) {
+        return NULL;
+    }
+    return &spice_audio_init;
+}
+
+static void spice_audio_fini (void *opaque)
+{
+    /* nothing */
+}
+
+static void rate_start (SpiceRateCtl *rate)
+{
+    memset (rate, 0, sizeof (*rate));
+    rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+}
+
+static int rate_get_samples (struct audio_pcm_info *info, SpiceRateCtl *rate)
+{
+    int64_t now;
+    int64_t ticks;
+    int64_t bytes;
+    int64_t samples;
+
+    now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+    ticks = now - rate->start_ticks;
+    bytes = muldiv64 (ticks, info->bytes_per_second, get_ticks_per_sec ());
+    samples = (bytes - rate->bytes_sent) >> info->shift;
+    if (samples < 0 || samples > 65536) {
+        error_report("Resetting rate control (%" PRId64 " samples)", samples);
+        rate_start (rate);
+        samples = 0;
+    }
+    rate->bytes_sent += samples << info->shift;
+    return samples;
+}
+
+/* playback */
+
+static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
+                         void *drv_opaque)
+{
+    SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
+    struct audsettings settings;
+
+#if SPICE_INTERFACE_PLAYBACK_MAJOR > 1 || SPICE_INTERFACE_PLAYBACK_MINOR >= 3
+    settings.freq       = spice_server_get_best_playback_rate(NULL);
+#else
+    settings.freq       = SPICE_INTERFACE_PLAYBACK_FREQ;
+#endif
+    settings.nchannels  = SPICE_INTERFACE_PLAYBACK_CHAN;
+    settings.fmt        = AUD_FMT_S16;
+    settings.endianness = AUDIO_HOST_ENDIANNESS;
+
+    audio_pcm_init_info (&hw->info, &settings);
+    hw->samples = LINE_OUT_SAMPLES;
+    out->active = 0;
+
+    out->sin.base.sif = &playback_sif.base;
+    qemu_spice_add_interface (&out->sin.base);
+#if SPICE_INTERFACE_PLAYBACK_MAJOR > 1 || SPICE_INTERFACE_PLAYBACK_MINOR >= 3
+    spice_server_set_playback_rate(&out->sin, settings.freq);
+#endif
+    return 0;
+}
+
+static void line_out_fini (HWVoiceOut *hw)
+{
+    SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
+
+    spice_server_remove_interface (&out->sin.base);
+}
+
+static int line_out_run (HWVoiceOut *hw, int live)
+{
+    SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
+    int rpos, decr;
+    int samples;
+
+    if (!live) {
+        return 0;
+    }
+
+    decr = rate_get_samples (&hw->info, &out->rate);
+    decr = audio_MIN (live, decr);
+
+    samples = decr;
+    rpos = hw->rpos;
+    while (samples) {
+        int left_till_end_samples = hw->samples - rpos;
+        int len = audio_MIN (samples, left_till_end_samples);
+
+        if (!out->frame) {
+            spice_server_playback_get_buffer (&out->sin, &out->frame, &out->fsize);
+            out->fpos = out->frame;
+        }
+        if (out->frame) {
+            len = audio_MIN (len, out->fsize);
+            hw->clip (out->fpos, hw->mix_buf + rpos, len);
+            out->fsize -= len;
+            out->fpos  += len;
+            if (out->fsize == 0) {
+                spice_server_playback_put_samples (&out->sin, out->frame);
+                out->frame = out->fpos = NULL;
+            }
+        }
+        rpos = (rpos + len) % hw->samples;
+        samples -= len;
+    }
+    hw->rpos = rpos;
+    return decr;
+}
+
+static int line_out_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
+{
+    SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        if (out->active) {
+            break;
+        }
+        out->active = 1;
+        rate_start (&out->rate);
+        spice_server_playback_start (&out->sin);
+        break;
+    case VOICE_DISABLE:
+        if (!out->active) {
+            break;
+        }
+        out->active = 0;
+        if (out->frame) {
+            memset (out->fpos, 0, out->fsize << 2);
+            spice_server_playback_put_samples (&out->sin, out->frame);
+            out->frame = out->fpos = NULL;
+        }
+        spice_server_playback_stop (&out->sin);
+        break;
+    case VOICE_VOLUME:
+        {
+#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
+            SWVoiceOut *sw;
+            va_list ap;
+            uint16_t vol[2];
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceOut *);
+            va_end (ap);
+
+            vol[0] = sw->vol.l / ((1ULL << 16) + 1);
+            vol[1] = sw->vol.r / ((1ULL << 16) + 1);
+            spice_server_playback_set_volume (&out->sin, 2, vol);
+            spice_server_playback_set_mute (&out->sin, sw->vol.mute);
+#endif
+            break;
+        }
+    }
+
+    return 0;
+}
+
+/* record */
+
+static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+    SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
+    struct audsettings settings;
+
+#if SPICE_INTERFACE_RECORD_MAJOR > 2 || SPICE_INTERFACE_RECORD_MINOR >= 3
+    settings.freq       = spice_server_get_best_record_rate(NULL);
+#else
+    settings.freq       = SPICE_INTERFACE_RECORD_FREQ;
+#endif
+    settings.nchannels  = SPICE_INTERFACE_RECORD_CHAN;
+    settings.fmt        = AUD_FMT_S16;
+    settings.endianness = AUDIO_HOST_ENDIANNESS;
+
+    audio_pcm_init_info (&hw->info, &settings);
+    hw->samples = LINE_IN_SAMPLES;
+    in->active = 0;
+
+    in->sin.base.sif = &record_sif.base;
+    qemu_spice_add_interface (&in->sin.base);
+#if SPICE_INTERFACE_RECORD_MAJOR > 2 || SPICE_INTERFACE_RECORD_MINOR >= 3
+    spice_server_set_record_rate(&in->sin, settings.freq);
+#endif
+    return 0;
+}
+
+static void line_in_fini (HWVoiceIn *hw)
+{
+    SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
+
+    spice_server_remove_interface (&in->sin.base);
+}
+
+static int line_in_run (HWVoiceIn *hw)
+{
+    SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
+    int num_samples;
+    int ready;
+    int len[2];
+    uint64_t delta_samp;
+    const uint32_t *samples;
+
+    if (!(num_samples = hw->samples - audio_pcm_hw_get_live_in (hw))) {
+        return 0;
+    }
+
+    delta_samp = rate_get_samples (&hw->info, &in->rate);
+    num_samples = audio_MIN (num_samples, delta_samp);
+
+    ready = spice_server_record_get_samples (&in->sin, in->samples, num_samples);
+    samples = in->samples;
+    if (ready == 0) {
+        static const uint32_t silence[LINE_IN_SAMPLES];
+        samples = silence;
+        ready = LINE_IN_SAMPLES;
+    }
+
+    num_samples = audio_MIN (ready, num_samples);
+
+    if (hw->wpos + num_samples > hw->samples) {
+        len[0] = hw->samples - hw->wpos;
+        len[1] = num_samples - len[0];
+    } else {
+        len[0] = num_samples;
+        len[1] = 0;
+    }
+
+    hw->conv (hw->conv_buf + hw->wpos, samples, len[0]);
+
+    if (len[1]) {
+        hw->conv (hw->conv_buf, samples + len[0], len[1]);
+    }
+
+    hw->wpos = (hw->wpos + num_samples) % hw->samples;
+
+    return num_samples;
+}
+
+static int line_in_read (SWVoiceIn *sw, void *buf, int size)
+{
+    return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
+{
+    SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        if (in->active) {
+            break;
+        }
+        in->active = 1;
+        rate_start (&in->rate);
+        spice_server_record_start (&in->sin);
+        break;
+    case VOICE_DISABLE:
+        if (!in->active) {
+            break;
+        }
+        in->active = 0;
+        spice_server_record_stop (&in->sin);
+        break;
+    case VOICE_VOLUME:
+        {
+#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
+            SWVoiceIn *sw;
+            va_list ap;
+            uint16_t vol[2];
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceIn *);
+            va_end (ap);
+
+            vol[0] = sw->vol.l / ((1ULL << 16) + 1);
+            vol[1] = sw->vol.r / ((1ULL << 16) + 1);
+            spice_server_record_set_volume (&in->sin, 2, vol);
+            spice_server_record_set_mute (&in->sin, sw->vol.mute);
+#endif
+            break;
+        }
+    }
+
+    return 0;
+}
+
+static struct audio_option audio_options[] = {
+    { /* end of list */ },
+};
+
+static struct audio_pcm_ops audio_callbacks = {
+    .init_out = line_out_init,
+    .fini_out = line_out_fini,
+    .run_out  = line_out_run,
+    .write    = line_out_write,
+    .ctl_out  = line_out_ctl,
+
+    .init_in  = line_in_init,
+    .fini_in  = line_in_fini,
+    .run_in   = line_in_run,
+    .read     = line_in_read,
+    .ctl_in   = line_in_ctl,
+};
+
+struct audio_driver spice_audio_driver = {
+    .name           = "spice",
+    .descr          = "spice audio driver",
+    .options        = audio_options,
+    .init           = spice_audio_init,
+    .fini           = spice_audio_fini,
+    .pcm_ops        = &audio_callbacks,
+    .max_voices_out = 1,
+    .max_voices_in  = 1,
+    .voice_size_out = sizeof (SpiceVoiceOut),
+    .voice_size_in  = sizeof (SpiceVoiceIn),
+#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
+    .ctl_caps       = VOICE_VOLUME_CAP
+#endif
+};
+
+void qemu_spice_audio_init (void)
+{
+    spice_audio_driver.can_be_default = 1;
+}
diff --git a/qemu/audio/wavaudio.c b/qemu/audio/wavaudio.c
new file mode 100644
index 000000000..c586020c5
--- /dev/null
+++ b/qemu/audio/wavaudio.c
@@ -0,0 +1,292 @@
+/*
+ * QEMU WAV 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 "hw/hw.h"
+#include "qemu/timer.h"
+#include "audio.h"
+
+#define AUDIO_CAP "wav"
+#include "audio_int.h"
+
+typedef struct WAVVoiceOut {
+    HWVoiceOut hw;
+    FILE *f;
+    int64_t old_ticks;
+    void *pcm_buf;
+    int total_samples;
+} WAVVoiceOut;
+
+typedef struct {
+    struct audsettings settings;
+    const char *wav_path;
+} WAVConf;
+
+static int wav_run_out (HWVoiceOut *hw, int live)
+{
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+    int rpos, decr, samples;
+    uint8_t *dst;
+    struct st_sample *src;
+    int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+    int64_t ticks = now - wav->old_ticks;
+    int64_t bytes =
+        muldiv64 (ticks, hw->info.bytes_per_second, get_ticks_per_sec ());
+
+    if (bytes > INT_MAX) {
+        samples = INT_MAX >> hw->info.shift;
+    }
+    else {
+        samples = bytes >> hw->info.shift;
+    }
+
+    wav->old_ticks = now;
+    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 = hw->mix_buf + rpos;
+        dst = advance (wav->pcm_buf, rpos << hw->info.shift);
+
+        hw->clip (dst, src, convert_samples);
+        if (fwrite (dst, convert_samples << hw->info.shift, 1, wav->f) != 1) {
+            dolog ("wav_run_out: fwrite of %d bytes failed\nReaons: %s\n",
+                   convert_samples << hw->info.shift, strerror (errno));
+        }
+
+        rpos = (rpos + convert_samples) % hw->samples;
+        samples -= convert_samples;
+        wav->total_samples += convert_samples;
+    }
+
+    hw->rpos = rpos;
+    return decr;
+}
+
+static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_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_init_out(HWVoiceOut *hw, struct audsettings *as,
+                        void *drv_opaque)
+{
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+    int bits16 = 0, stereo = 0;
+    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
+    };
+    WAVConf *conf = drv_opaque;
+    struct audsettings wav_as = conf->settings;
+
+    stereo = wav_as.nchannels == 2;
+    switch (wav_as.fmt) {
+    case AUD_FMT_S8:
+    case AUD_FMT_U8:
+        bits16 = 0;
+        break;
+
+    case AUD_FMT_S16:
+    case AUD_FMT_U16:
+        bits16 = 1;
+        break;
+
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        dolog ("WAVE files can not handle 32bit formats\n");
+        return -1;
+    }
+
+    hdr[34] = bits16 ? 0x10 : 0x08;
+
+    wav_as.endianness = 0;
+    audio_pcm_init_info (&hw->info, &wav_as);
+
+    hw->samples = 1024;
+    wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    if (!wav->pcm_buf) {
+        dolog ("Could not allocate buffer (%d bytes)\n",
+               hw->samples << hw->info.shift);
+        return -1;
+    }
+
+    le_store (hdr + 22, hw->info.nchannels, 2);
+    le_store (hdr + 24, hw->info.freq, 4);
+    le_store (hdr + 28, hw->info.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));
+        g_free (wav->pcm_buf);
+        wav->pcm_buf = NULL;
+        return -1;
+    }
+
+    if (fwrite (hdr, sizeof (hdr), 1, wav->f) != 1) {
+        dolog ("wav_init_out: failed to write header\nReason: %s\n",
+               strerror(errno));
+        return -1;
+    }
+    return 0;
+}
+
+static void wav_fini_out (HWVoiceOut *hw)
+{
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+    uint8_t rlen[4];
+    uint8_t dlen[4];
+    uint32_t datalen = wav->total_samples << hw->info.shift;
+    uint32_t rifflen = datalen + 36;
+
+    if (!wav->f) {
+        return;
+    }
+
+    le_store (rlen, rifflen, 4);
+    le_store (dlen, datalen, 4);
+
+    if (fseek (wav->f, 4, SEEK_SET)) {
+        dolog ("wav_fini_out: fseek to rlen failed\nReason: %s\n",
+               strerror(errno));
+        goto doclose;
+    }
+    if (fwrite (rlen, 4, 1, wav->f) != 1) {
+        dolog ("wav_fini_out: failed to write rlen\nReason: %s\n",
+               strerror (errno));
+        goto doclose;
+    }
+    if (fseek (wav->f, 32, SEEK_CUR)) {
+        dolog ("wav_fini_out: fseek to dlen failed\nReason: %s\n",
+               strerror (errno));
+        goto doclose;
+    }
+    if (fwrite (dlen, 4, 1, wav->f) != 1) {
+        dolog ("wav_fini_out: failed to write dlen\nReaons: %s\n",
+               strerror (errno));
+        goto doclose;
+    }
+
+ doclose:
+    if (fclose (wav->f))  {
+        dolog ("wav_fini_out: fclose %p failed\nReason: %s\n",
+               wav->f, strerror (errno));
+    }
+    wav->f = NULL;
+
+    g_free (wav->pcm_buf);
+    wav->pcm_buf = NULL;
+}
+
+static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static WAVConf glob_conf = {
+    .settings.freq      = 44100,
+    .settings.nchannels = 2,
+    .settings.fmt       = AUD_FMT_S16,
+    .wav_path           = "qemu.wav"
+};
+
+static void *wav_audio_init (void)
+{
+    WAVConf *conf = g_malloc(sizeof(WAVConf));
+    *conf = glob_conf;
+    return conf;
+}
+
+static void wav_audio_fini (void *opaque)
+{
+    ldebug ("wav_fini");
+    g_free(opaque);
+}
+
+static struct audio_option wav_options[] = {
+    {
+        .name  = "FREQUENCY",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.settings.freq,
+        .descr = "Frequency"
+    },
+    {
+        .name  = "FORMAT",
+        .tag   = AUD_OPT_FMT,
+        .valp  = &glob_conf.settings.fmt,
+        .descr = "Format"
+    },
+    {
+        .name  = "DAC_FIXED_CHANNELS",
+        .tag   = AUD_OPT_INT,
+        .valp  = &glob_conf.settings.nchannels,
+        .descr = "Number of channels (1 - mono, 2 - stereo)"
+    },
+    {
+        .name  = "PATH",
+        .tag   = AUD_OPT_STR,
+        .valp  = &glob_conf.wav_path,
+        .descr = "Path to wave file"
+    },
+    { /* End of list */ }
+};
+
+static struct audio_pcm_ops wav_pcm_ops = {
+    .init_out = wav_init_out,
+    .fini_out = wav_fini_out,
+    .run_out  = wav_run_out,
+    .write    = wav_write_out,
+    .ctl_out  = wav_ctl_out,
+};
+
+struct audio_driver wav_audio_driver = {
+    .name           = "wav",
+    .descr          = "WAV renderer http://wikipedia.org/wiki/WAV",
+    .options        = wav_options,
+    .init           = wav_audio_init,
+    .fini           = wav_audio_fini,
+    .pcm_ops        = &wav_pcm_ops,
+    .can_be_default = 0,
+    .max_voices_out = 1,
+    .max_voices_in  = 0,
+    .voice_size_out = sizeof (WAVVoiceOut),
+    .voice_size_in  = 0
+};
diff --git a/qemu/audio/wavcapture.c b/qemu/audio/wavcapture.c
new file mode 100644
index 000000000..86e905627
--- /dev/null
+++ b/qemu/audio/wavcapture.c
@@ -0,0 +1,194 @@
+#include "hw/hw.h"
+#include "monitor/monitor.h"
+#include "qemu/error-report.h"
+#include "audio.h"
+
+typedef struct {
+    FILE *f;
+    int bytes;
+    char *path;
+    int freq;
+    int bits;
+    int nchannels;
+    CaptureVoiceOut *cap;
+} WAVState;
+
+/* 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 void wav_notify (void *opaque, audcnotification_e cmd)
+{
+    (void) opaque;
+    (void) cmd;
+}
+
+static void wav_destroy (void *opaque)
+{
+    WAVState *wav = opaque;
+    uint8_t rlen[4];
+    uint8_t dlen[4];
+    uint32_t datalen = wav->bytes;
+    uint32_t rifflen = datalen + 36;
+    Monitor *mon = cur_mon;
+
+    if (wav->f) {
+        le_store (rlen, rifflen, 4);
+        le_store (dlen, datalen, 4);
+
+        if (fseek (wav->f, 4, SEEK_SET)) {
+            monitor_printf (mon, "wav_destroy: rlen fseek failed\nReason: %s\n",
+                            strerror (errno));
+            goto doclose;
+        }
+        if (fwrite (rlen, 4, 1, wav->f) != 1) {
+            monitor_printf (mon, "wav_destroy: rlen fwrite failed\nReason %s\n",
+                            strerror (errno));
+            goto doclose;
+        }
+        if (fseek (wav->f, 32, SEEK_CUR)) {
+            monitor_printf (mon, "wav_destroy: dlen fseek failed\nReason %s\n",
+                            strerror (errno));
+            goto doclose;
+        }
+        if (fwrite (dlen, 1, 4, wav->f) != 4) {
+            monitor_printf (mon, "wav_destroy: dlen fwrite failed\nReason %s\n",
+                            strerror (errno));
+            goto doclose;
+        }
+    doclose:
+        if (fclose (wav->f)) {
+            error_report("wav_destroy: fclose failed: %s", strerror(errno));
+        }
+    }
+
+    g_free (wav->path);
+}
+
+static void wav_capture (void *opaque, void *buf, int size)
+{
+    WAVState *wav = opaque;
+
+    if (fwrite (buf, size, 1, wav->f) != 1) {
+        monitor_printf (cur_mon, "wav_capture: fwrite error\nReason: %s",
+                        strerror (errno));
+    }
+    wav->bytes += size;
+}
+
+static void wav_capture_destroy (void *opaque)
+{
+    WAVState *wav = opaque;
+
+    AUD_del_capture (wav->cap, wav);
+}
+
+static void wav_capture_info (void *opaque)
+{
+    WAVState *wav = opaque;
+    char *path = wav->path;
+
+    monitor_printf (cur_mon, "Capturing audio(%d,%d,%d) to %s: %d bytes\n",
+                    wav->freq, wav->bits, wav->nchannels,
+                    path ? path : "<not available>", wav->bytes);
+}
+
+static struct capture_ops wav_capture_ops = {
+    .destroy = wav_capture_destroy,
+    .info = wav_capture_info
+};
+
+int wav_start_capture (CaptureState *s, const char *path, int freq,
+                       int bits, int nchannels)
+{
+    Monitor *mon = cur_mon;
+    WAVState *wav;
+    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
+    };
+    struct audsettings as;
+    struct audio_capture_ops ops;
+    int stereo, bits16, shift;
+    CaptureVoiceOut *cap;
+
+    if (bits != 8 && bits != 16) {
+        monitor_printf (mon, "incorrect bit count %d, must be 8 or 16\n", bits);
+        return -1;
+    }
+
+    if (nchannels != 1 && nchannels != 2) {
+        monitor_printf (mon, "incorrect channel count %d, must be 1 or 2\n",
+                        nchannels);
+        return -1;
+    }
+
+    stereo = nchannels == 2;
+    bits16 = bits == 16;
+
+    as.freq = freq;
+    as.nchannels = 1 << stereo;
+    as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+    as.endianness = 0;
+
+    ops.notify = wav_notify;
+    ops.capture = wav_capture;
+    ops.destroy = wav_destroy;
+
+    wav = g_malloc0 (sizeof (*wav));
+
+    shift = bits16 + stereo;
+    hdr[34] = bits16 ? 0x10 : 0x08;
+
+    le_store (hdr + 22, as.nchannels, 2);
+    le_store (hdr + 24, freq, 4);
+    le_store (hdr + 28, freq << shift, 4);
+    le_store (hdr + 32, 1 << shift, 2);
+
+    wav->f = fopen (path, "wb");
+    if (!wav->f) {
+        monitor_printf (mon, "Failed to open wave file `%s'\nReason: %s\n",
+                        path, strerror (errno));
+        g_free (wav);
+        return -1;
+    }
+
+    wav->path = g_strdup (path);
+    wav->bits = bits;
+    wav->nchannels = nchannels;
+    wav->freq = freq;
+
+    if (fwrite (hdr, sizeof (hdr), 1, wav->f) != 1) {
+        monitor_printf (mon, "Failed to write header\nReason: %s\n",
+                        strerror (errno));
+        goto error_free;
+    }
+
+    cap = AUD_add_capture (&as, &ops, wav);
+    if (!cap) {
+        monitor_printf (mon, "Failed to add audio capture\n");
+        goto error_free;
+    }
+
+    wav->cap = cap;
+    s->opaque = wav;
+    s->ops = wav_capture_ops;
+    return 0;
+
+error_free:
+    g_free (wav->path);
+    if (fclose (wav->f)) {
+        monitor_printf (mon, "Failed to close wave file\nReason: %s\n",
+                        strerror (errno));
+    }
+    g_free (wav);
+    return -1;
+}