18 files changed, 6609 insertions, 1371 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
new file mode 100644
index 0000000000..133690576e
--- /dev/null
+++ b/audio/alsaaudio.c
@@ -0,0 +1,926 @@
+/*
+ * 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 "vl.h"
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAVoiceOut {
+    HWVoiceOut hw;
+    void *pcm_buf;
+    snd_pcm_t *handle;
+    int can_pause;
+    int was_enabled;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+    HWVoiceIn hw;
+    snd_pcm_t *handle;
+    void *pcm_buf;
+    int can_pause;
+} ALSAVoiceIn;
+
+static struct {
+    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_overriden;
+    int period_size_in_overriden;
+
+    int buffer_size_out_overriden;
+    int period_size_out_overriden;
+} conf = {
+#ifdef HIGH_LATENCY
+    .size_in_usec_in = 1,
+    .size_in_usec_out = 1,
+#endif
+    .pcm_name_out = "hw:0,0",
+    .pcm_name_in = "hw:0,0",
+#ifdef HIGH_LATENCY
+    .buffer_size_in = 400000,
+    .period_size_in = 400000 / 4,
+    .buffer_size_out = 400000,
+    .period_size_out = 400000 / 4,
+#else
+#define DEFAULT_BUFFER_SIZE 1024
+#define DEFAULT_PERIOD_SIZE 256
+    .buffer_size_in = DEFAULT_BUFFER_SIZE,
+    .period_size_in = DEFAULT_PERIOD_SIZE,
+    .buffer_size_out = DEFAULT_BUFFER_SIZE,
+    .period_size_out = DEFAULT_PERIOD_SIZE,
+    .buffer_size_in_overriden = 0,
+    .buffer_size_out_overriden = 0,
+    .period_size_in_overriden = 0,
+    .period_size_out_overriden = 0,
+#endif
+    .threshold = 0
+};
+
+struct alsa_params_req {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    unsigned int buffer_size;
+    unsigned int period_size;
+};
+
+struct alsa_params_obt {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    int can_pause;
+    snd_pcm_uframes_t buffer_size;
+};
+
+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, "Can 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_anal_close (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 int alsa_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int aud_to_alsafmt (audfmt_e fmt)
+{
+    switch (fmt) {
+    case AUD_FMT_S8:
+        return SND_PCM_FORMAT_S8;
+
+    case AUD_FMT_U8:
+        return SND_PCM_FORMAT_U8;
+
+    case AUD_FMT_S16:
+        return SND_PCM_FORMAT_S16_LE;
+
+    case AUD_FMT_U16:
+        return SND_PCM_FORMAT_U16_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 (int 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;
+
+    default:
+        dolog ("Unrecognized audio format %d\n", alsafmt);
+        return -1;
+    }
+
+    return 0;
+}
+
+#ifdef DEBUG_MISMATCHES
+static void alsa_dump_info (struct alsa_params_req *req,
+                            struct alsa_params_obt *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 ("============================================\n");
+    dolog ("requested: buffer size %d period size %d\n",
+           req->buffer_size, req->period_size);
+    dolog ("obtained: buffer size %ld\n", obt->buffer_size);
+}
+#endif
+
+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 ("Can 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 ("Can 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 ("Can 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)
+{
+    snd_pcm_t *handle;
+    snd_pcm_hw_params_t *hw_params;
+    int err, freq, nchannels;
+    const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+    unsigned int period_size, buffer_size;
+    snd_pcm_uframes_t obt_buffer_size;
+    const char *typ = in ? "ADC" : "DAC";
+
+    freq = req->freq;
+    period_size = req->period_size;
+    buffer_size = req->buffer_size;
+    nchannels = req->nchannels;
+
+    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);
+        goto err;
+    }
+
+    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 (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
+        if (!buffer_size) {
+            buffer_size = DEFAULT_BUFFER_SIZE;
+            period_size= DEFAULT_PERIOD_SIZE;
+        }
+    }
+
+    if (buffer_size) {
+        if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
+            if (period_size) {
+                err = snd_pcm_hw_params_set_period_time_near (
+                    handle,
+                    hw_params,
+                    &period_size,
+                    0);
+                if (err < 0) {
+                    alsa_logerr2 (err, typ,
+                                  "Failed to set period time %d\n",
+                                  req->period_size);
+                    goto err;
+                }
+            }
+
+            err = snd_pcm_hw_params_set_buffer_time_near (
+                handle,
+                hw_params,
+                &buffer_size,
+                0);
+
+            if (err < 0) {
+                alsa_logerr2 (err, typ,
+                              "Failed to set buffer time %d\n",
+                              req->buffer_size);
+                goto err;
+            }
+        }
+        else {
+            int dir;
+            snd_pcm_uframes_t minval;
+
+            if (period_size) {
+                minval = period_size;
+                dir = 0;
+
+                err = snd_pcm_hw_params_get_period_size_min (
+                    hw_params,
+                    &minval,
+                    &dir
+                    );
+                if (err < 0) {
+                    alsa_logerr (
+                        err,
+                        "Can not get minmal period size for %s\n",
+                        typ
+                        );
+                }
+                else {
+                    if (period_size < minval) {
+                        if ((in && conf.period_size_in_overriden)
+                            || (!in && conf.period_size_out_overriden)) {
+                            dolog ("%s period size(%d) is less "
+                                   "than minmal period size(%ld)\n",
+                                   typ,
+                                   period_size,
+                                   minval);
+                        }
+                        period_size = minval;
+                    }
+                }
+
+                err = snd_pcm_hw_params_set_period_size (
+                    handle,
+                    hw_params,
+                    period_size,
+                    0
+                    );
+                if (err < 0) {
+                    alsa_logerr2 (err, typ, "Failed to set period size %d\n",
+                                  req->period_size);
+                    goto err;
+                }
+            }
+
+            minval = buffer_size;
+            err = snd_pcm_hw_params_get_buffer_size_min (
+                hw_params,
+                &minval
+                );
+            if (err < 0) {
+                alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+                             typ);
+            }
+            else {
+                if (buffer_size < minval) {
+                    if ((in && conf.buffer_size_in_overriden)
+                        || (!in && conf.buffer_size_out_overriden)) {
+                        dolog (
+                            "%s buffer size(%d) is less "
+                            "than minimal buffer size(%ld)\n",
+                            typ,
+                            buffer_size,
+                            minval
+                            );
+                    }
+                    buffer_size = minval;
+                }
+            }
+
+            err = snd_pcm_hw_params_set_buffer_size (
+                handle,
+                hw_params,
+                buffer_size
+                );
+            if (err < 0) {
+                alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
+                              req->buffer_size);
+                goto err;
+            }
+        }
+    }
+    else {
+        dolog ("warning: buffer size is not set\n");
+    }
+
+    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_prepare (handle);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+        goto err;
+    }
+
+    obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
+    if (obt->can_pause < 0) {
+        alsa_logerr (err, "Can not get pause capability for %s\n", typ);
+        obt->can_pause = 0;
+    }
+
+    if (!in && conf.threshold) {
+        snd_pcm_uframes_t threshold;
+        int bytes_per_sec;
+
+        bytes_per_sec = freq
+            << (nchannels == 2)
+            << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
+
+        threshold = (conf.threshold * bytes_per_sec) / 1000;
+        alsa_set_threshold (handle, threshold);
+    }
+
+    obt->fmt = req->fmt;
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+    *handlep = handle;
+
+    if (obt->fmt != req->fmt ||
+        obt->nchannels != req->nchannels ||
+        obt->freq != req->freq) {
+#ifdef DEBUG_MISMATCHES
+        dolog ("Audio paramters mismatch for %s\n", typ);
+        alsa_dump_info (req, obt);
+#endif
+    }
+
+#ifdef DEBUG
+    alsa_dump_info (req, obt);
+#endif
+    return 0;
+
+ err:
+    alsa_anal_close (&handle);
+    return -1;
+}
+
+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_run_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    int rpos, live, decr;
+    int samples;
+    uint8_t *dst;
+    st_sample_t *src;
+    snd_pcm_sframes_t avail;
+
+    live = audio_pcm_hw_get_live_out (hw);
+    if (!live) {
+        return 0;
+    }
+
+    avail = snd_pcm_avail_update (alsa->handle);
+    if (avail < 0) {
+        if (avail == -EPIPE) {
+            if (!alsa_recover (alsa->handle)) {
+                avail = snd_pcm_avail_update (alsa->handle);
+                if (avail >= 0) {
+                    goto ok;
+                }
+            }
+        }
+
+        alsa_logerr (avail, "Can not get amount free space\n");
+        return 0;
+    }
+
+ ok:
+    decr = audio_MIN (live, avail);
+    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);
+        snd_pcm_sframes_t written;
+
+        src = hw->mix_buf + rpos;
+        dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
+
+        hw->clip (dst, src, convert_samples);
+
+    again:
+        written = snd_pcm_writei (alsa->handle, dst, convert_samples);
+
+        if (written < 0) {
+            switch (written) {
+            case -EPIPE:
+                if (!alsa_recover (alsa->handle)) {
+                    goto again;
+                }
+                dolog (
+                    "Failed to write %d frames to %p, handle %p not prepared\n",
+                    convert_samples,
+                    dst,
+                    alsa->handle
+                    );
+                goto exit;
+
+            case -EAGAIN:
+                goto again;
+
+            default:
+                alsa_logerr (written, "Failed to write %d frames to %p\n",
+                             convert_samples, dst);
+                goto exit;
+            }
+        }
+
+        mixeng_clear (src, written);
+        rpos = (rpos + written) % hw->samples;
+        samples -= written;
+    }
+
+ exit:
+    hw->rpos = rpos;
+    return decr;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    ldebug ("alsa_fini\n");
+    alsa_anal_close (&alsa->handle);
+
+    if (alsa->pcm_buf) {
+        qemu_free (alsa->pcm_buf);
+        alsa->pcm_buf = NULL;
+    }
+}
+
+static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    audfmt_e effective_fmt;
+    int endianness;
+    int err;
+    snd_pcm_t *handle;
+
+    req.fmt = aud_to_alsafmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.period_size = conf.period_size_out;
+    req.buffer_size = conf.buffer_size_out;
+
+    if (alsa_open (0, &req, &obt, &handle)) {
+        return -1;
+    }
+
+    err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    alsa->can_pause = obt.can_pause;
+    hw->bufsize = obt.buffer_size;
+
+    alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!alsa->pcm_buf) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    alsa->was_enabled = 0;
+    return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    int err;
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        ldebug ("enabling voice\n");
+        audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples);
+        if (alsa->can_pause) {
+            /* Why this was_enabled madness is needed at all?? */
+            if (alsa->was_enabled) {
+                err = snd_pcm_pause (alsa->handle, 0);
+                if (err < 0) {
+                    alsa_logerr (err, "Failed to resume playing\n");
+                    /* not fatal really */
+                }
+            }
+            else {
+                alsa->was_enabled = 1;
+            }
+        }
+        break;
+
+    case VOICE_DISABLE:
+        ldebug ("disabling voice\n");
+        if (alsa->can_pause) {
+            err = snd_pcm_pause (alsa->handle, 1);
+            if (err < 0) {
+                alsa_logerr (err, "Failed to stop playing\n");
+                /* not fatal really */
+            }
+        }
+        break;
+    }
+    return 0;
+}
+
+static int alsa_init_in (HWVoiceIn *hw,
+                        int freq, int nchannels, audfmt_e fmt)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    int endianness;
+    int err;
+    audfmt_e effective_fmt;
+    snd_pcm_t *handle;
+
+    req.fmt = aud_to_alsafmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.period_size = conf.period_size_in;
+    req.buffer_size = conf.buffer_size_in;
+
+    if (alsa_open (1, &req, &obt, &handle)) {
+        return -1;
+    }
+
+    err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    alsa->can_pause = obt.can_pause;
+    hw->bufsize = obt.buffer_size;
+    alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!alsa->pcm_buf) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    alsa_anal_close (&alsa->handle);
+
+    if (alsa->pcm_buf) {
+        qemu_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;
+    struct {
+        int add;
+        int len;
+    } bufs[2] = {
+        { hw->wpos, 0 },
+        { 0, 0 }
+    };
+
+    snd_pcm_uframes_t read_samples = 0;
+
+    if (!dead) {
+        return 0;
+    }
+
+    if (hw->wpos + dead > hw->samples) {
+        bufs[0].len = (hw->samples - hw->wpos);
+        bufs[1].len = (dead - (hw->samples - hw->wpos));
+    }
+    else {
+        bufs[0].len = dead;
+    }
+
+
+    for (i = 0; i < 2; ++i) {
+        void *src;
+        st_sample_t *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 -EPIPE:
+                    if (!alsa_recover (alsa->handle)) {
+                        continue;
+                    }
+                    dolog (
+                        "Failed to read %ld frames from %p, "
+                        "handle %p not prepared\n",
+                        len,
+                        src,
+                        alsa->handle
+                        );
+                    goto exit;
+
+                case -EAGAIN:
+                    continue;
+
+                default:
+                    alsa_logerr (
+                        nread,
+                        "Failed to read %ld frames from %p\n",
+                        len,
+                        src
+                        );
+                    goto exit;
+                }
+            }
+
+            hw->conv (dst, src, nread, &nominal_volume);
+
+            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, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static void *alsa_audio_init (void)
+{
+    return &conf;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+    (void) opaque;
+}
+
+static struct audio_option alsa_options[] = {
+    {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
+     "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+    {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
+     "DAC period size", &conf.period_size_out_overriden, 0},
+    {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
+     "DAC buffer size", &conf.buffer_size_out_overriden, 0},
+
+    {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
+     "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+    {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
+     "ADC period size", &conf.period_size_in_overriden, 0},
+    {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
+     "ADC buffer size", &conf.buffer_size_in_overriden, 0},
+
+    {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+     "(undocumented)", NULL, 0},
+
+    {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
+     "DAC device name (for instance dmix)", NULL, 0},
+
+    {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
+     "ADC device name", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+    alsa_init_out,
+    alsa_fini_out,
+    alsa_run_out,
+    alsa_write,
+    alsa_ctl_out,
+
+    alsa_init_in,
+    alsa_fini_in,
+    alsa_run_in,
+    alsa_read,
+    alsa_ctl_in
+};
+
+struct audio_driver alsa_audio_driver = {
+    INIT_FIELD (name           = ) "alsa",
+    INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
+    INIT_FIELD (options        = ) alsa_options,
+    INIT_FIELD (init           = ) alsa_audio_init,
+    INIT_FIELD (fini           = ) alsa_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) INT_MAX,
+    INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
+};
diff --git a/audio/audio.c b/audio/audio.c
index 0c0c8dd86f..1a3925d4e9 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1,8 +1,8 @@
 /*
  * QEMU Audio subsystem
- * 
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- * 
+ *
+ * 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
@@ -21,34 +21,78 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
-#include <assert.h>
 #include "vl.h"
 
-#define USE_WAV_AUDIO
+#define AUDIO_CAP "audio"
+#include "audio_int.h"
 
-#include "audio/audio_int.h"
+static void audio_pcm_hw_fini_in (HWVoiceIn *hw);
+static void audio_pcm_hw_fini_out (HWVoiceOut *hw);
 
-#define dolog(...) AUD_log ("audio", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+static LIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
+static LIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
 
-#define QC_AUDIO_DRV    "QEMU_AUDIO_DRV"
-#define QC_VOICES       "QEMU_VOICES"
-#define QC_FIXED_FORMAT "QEMU_FIXED_FORMAT"
-#define QC_FIXED_FREQ   "QEMU_FIXED_FREQ"
+/* #define DEBUG_PLIVE */
+/* #define DEBUG_LIVE */
+/* #define DEBUG_OUT */
 
-static HWVoice *hw_voices;
+static struct audio_driver *drvtab[] = {
+#ifdef CONFIG_OSS
+    &oss_audio_driver,
+#endif
+#ifdef CONFIG_ALSA
+    &alsa_audio_driver,
+#endif
+#ifdef CONFIG_COREAUDIO
+    &coreaudio_audio_driver,
+#endif
+#ifdef CONFIG_DSOUND
+    &dsound_audio_driver,
+#endif
+#ifdef CONFIG_FMOD
+    &fmod_audio_driver,
+#endif
+#ifdef CONFIG_SDL
+    &sdl_audio_driver,
+#endif
+    &no_audio_driver,
+    &wav_audio_driver
+};
 
 AudioState audio_state = {
+    /* Out */
+    1,                          /* use fixed settings */
+    44100,                      /* fixed frequency */
+    2,                          /* fixed channels */
+    AUD_FMT_S16,                /* fixed format */
+    1,                          /* number of hw voices */
+    1,                          /* greedy */
+
+    /* In */
     1,                          /* use fixed settings */
     44100,                      /* fixed frequency */
     2,                          /* fixed channels */
     AUD_FMT_S16,                /* fixed format */
     1,                          /* number of hw voices */
-    -1                          /* voice size */
+    1,                          /* greedy */
+
+    NULL,                       /* driver opaque */
+    NULL,                       /* driver */
+
+    NULL,                       /* timer handle */
+    { 0 },                      /* period */
+    0                           /* plive */
+};
+
+volume_t nominal_volume = {
+    0,
+#ifdef FLOAT_MIXENG
+    1.0,
+    1.0
+#else
+    UINT_MAX,
+    UINT_MAX
+#endif
 };
 
 /* http://www.df.lth.se/~john_e/gems/gem002d.html */
@@ -68,70 +112,334 @@ inline uint32_t lsbindex (uint32_t u)
     return popcount ((u&-u)-1);
 }
 
-int audio_get_conf_int (const char *key, int defval)
+#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
+#error No its not
+#else
+int audio_bug (const char *funcname, int cond)
+{
+    if (cond) {
+        static int shown;
+
+        AUD_log (NULL, "Error a bug that was just triggered in %s\n", funcname);
+        if (!shown) {
+            shown = 1;
+            AUD_log (NULL, "Save all your work and restart without audio\n");
+            AUD_log (NULL, "Please send bug report to malc@pulsesoft.com\n");
+            AUD_log (NULL, "I am sorry\n");
+        }
+        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 char *audio_alloc_prefix (const char *s)
+{
+    const char qemu_prefix[] = "QEMU_";
+    size_t len;
+    char *r;
+
+    if (!s) {
+        return NULL;
+    }
+
+    len = strlen (s);
+    r = qemu_malloc (len + sizeof (qemu_prefix));
+
+    if (r) {
+        size_t i;
+        char *u = r + sizeof (qemu_prefix) - 1;
+
+        strcpy (r, qemu_prefix);
+        strcat (r, s);
+
+        for (i = 0; i < len; ++i) {
+            u[i] = toupper (u[i]);
+        }
+    }
+    return r;
+}
+
+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";
+    }
+
+    dolog ("Bogus audfmt %d returning S16\n", fmt);
+    return "S16";
+}
+
+audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
 {
-    int val = defval;
+    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, "s8")) {
+        *defaultp = 0;
+        return AUD_FMT_S8;
+    }
+    else if (!strcasecmp (s, "s16")) {
+        *defaultp = 0;
+        return AUD_FMT_S16;
+    }
+    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;
     }
-
-    return val;
 }
 
-const char *audio_get_conf_str (const char *key, const char *defval)
+static const char *audio_get_conf_str (const char *key,
+                                       const char *defval,
+                                       int *defaultp)
 {
     const char *val = getenv (key);
-    if (!val)
+    if (!val) {
+        *defaultp = 1;
         return defval;
-    else
+    }
+    else {
+        *defaultp = 0;
         return val;
+    }
 }
 
 void AUD_log (const char *cap, const char *fmt, ...)
 {
     va_list ap;
-    fprintf (stderr, "%s: ", cap);
+    if (cap) {
+        fprintf (stderr, "%s: ", cap);
+    }
     va_start (ap, fmt);
     vfprintf (stderr, fmt, ap);
     va_end (ap);
 }
 
-/*
- * Soft Voice
- */
-void pcm_sw_free_resources (SWVoice *sw)
+void AUD_vlog (const char *cap, const char *fmt, va_list ap)
 {
-    if (sw->buf) qemu_free (sw->buf);
-    if (sw->rate) st_rate_stop (sw->rate);
-    sw->buf = NULL;
-    sw->rate = NULL;
+    if (cap) {
+        fprintf (stderr, "%s: ", cap);
+    }
+    vfprintf (stderr, fmt, ap);
 }
 
-int pcm_sw_alloc_resources (SWVoice *sw)
+static void audio_print_options (const char *prefix,
+                                 struct audio_option *opt)
 {
-    sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t));
-    if (!sw->buf)
-        return -1;
+    char *uprefix;
 
-    sw->rate = st_rate_start (sw->freq, sw->hw->freq);
-    if (!sw->rate) {
-        qemu_free (sw->buf);
-        sw->buf = NULL;
-        return -1;
+    if (!prefix) {
+        dolog ("No prefix specified\n");
+        return;
     }
-    return 0;
+
+    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->overridenp && *opt->overridenp) {
+            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)\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);
+    }
+
+    qemu_free (uprefix);
 }
 
-void pcm_sw_fini (SWVoice *sw)
+static void audio_process_options (const char *prefix,
+                                   struct audio_option *opt)
 {
-    pcm_sw_free_resources (sw);
+    char *optname;
+    const char qemu_prefix[] = "QEMU_";
+    size_t preflen;
+
+    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);
+        optname = qemu_malloc (len + preflen + sizeof (qemu_prefix) + 1);
+        if (!optname) {
+            dolog ("Can not allocate memory for option name `%s'\n",
+                   opt->name);
+            continue;
+        }
+
+        strcpy (optname, qemu_prefix);
+        for (i = 0; i <= preflen; ++i) {
+            optname[i + sizeof (qemu_prefix) - 1] = toupper (prefix[i]);
+        }
+        strcat (optname, "_");
+        strcat (optname, 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->overridenp) {
+            opt->overridenp = &opt->overriden;
+        }
+        *opt->overridenp = !def;
+        qemu_free (optname);
+    }
 }
 
-int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
-                 int nchannels, audfmt_e fmt)
+static int audio_pcm_info_eq (struct audio_pcm_info *info, int freq,
+                              int nchannels, audfmt_e fmt)
 {
     int bits = 8, sign = 0;
 
@@ -147,746 +455,1167 @@ int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
         bits = 16;
         break;
     }
+    return info->freq == freq
+        && info->nchannels == nchannels
+        && info->sign == sign
+        && info->bits == bits;
+}
 
-    sw->hw = hw;
-    sw->freq = freq;
-    sw->fmt = fmt;
-    sw->nchannels = nchannels;
-    sw->shift = (nchannels == 2) + (bits == 16);
-    sw->align = (1 << sw->shift) - 1;
-    sw->left = 0;
-    sw->pos = 0;
-    sw->wpos = 0;
-    sw->live = 0;
-    sw->ratio = (sw->hw->freq * ((int64_t) INT_MAX)) / sw->freq;
-    sw->bytes_per_second = sw->freq << sw->shift;
-    sw->conv = mixeng_conv[nchannels == 2][sign][bits == 16];
-
-    pcm_sw_free_resources (sw);
-    return pcm_sw_alloc_resources (sw);
-}
-
-/* Hard voice */
-void pcm_hw_free_resources (HWVoice *hw)
-{
-    if (hw->mix_buf)
-        qemu_free (hw->mix_buf);
-    hw->mix_buf = NULL;
+void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
+                          int nchannels, audfmt_e fmt, int swap_endian)
+{
+    int bits = 8, sign = 0;
+
+    switch (fmt) {
+    case AUD_FMT_S8:
+        sign = 1;
+    case AUD_FMT_U8:
+        break;
+
+    case AUD_FMT_S16:
+        sign = 1;
+    case AUD_FMT_U16:
+        bits = 16;
+        break;
+    }
+
+    info->freq = freq;
+    info->bits = bits;
+    info->sign = sign;
+    info->nchannels = nchannels;
+    info->shift = (nchannels == 2) + (bits == 16);
+    info->align = (1 << info->shift) - 1;
+    info->bytes_per_second = info->freq << info->shift;
+    info->swap_endian = swap_endian;
 }
 
-int pcm_hw_alloc_resources (HWVoice *hw)
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
 {
-    hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
-    if (!hw->mix_buf)
-        return -1;
-    return 0;
+    if (!len) {
+        return;
+    }
+
+    if (info->sign) {
+        memset (buf, len << info->shift, 0x00);
+    }
+    else {
+        if (info->bits == 8) {
+            memset (buf, len << info->shift, 0x80);
+        }
+        else {
+            int i;
+            uint16_t *p = buf;
+            int shift = info->nchannels - 1;
+            short s = INT16_MAX;
+
+            if (info->swap_endian) {
+                s = bswap16 (s);
+            }
+
+            for (i = 0; i < len << shift; i++) {
+                p[i] = s;
+            }
+        }
+    }
 }
 
-void pcm_hw_fini (HWVoice *hw)
+/*
+ * Hard voice (capture)
+ */
+static void audio_pcm_hw_free_resources_in (HWVoiceIn *hw)
 {
-    if (hw->active) {
-        ldebug ("pcm_hw_fini: %d %d %d\n", hw->freq, hw->nchannels, hw->fmt);
-        pcm_hw_free_resources (hw);
-        hw->pcm_ops->fini (hw);
-        memset (hw, 0, audio_state.drv->voice_size);
+    if (hw->conv_buf) {
+        qemu_free (hw->conv_buf);
     }
+    hw->conv_buf = NULL;
 }
 
-void pcm_hw_gc (HWVoice *hw)
+static int audio_pcm_hw_alloc_resources_in (HWVoiceIn *hw)
 {
-    if (hw->nb_voices)
-        return;
+    hw->conv_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
+    if (!hw->conv_buf) {
+        return -1;
+    }
+    return 0;
+}
+
+static int audio_pcm_hw_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    audio_pcm_hw_fini_in (hw);
 
-    pcm_hw_fini (hw);
+    if (hw->pcm_ops->init_in (hw, freq, nchannels, fmt)) {
+        memset (hw, 0, audio_state.drv->voice_size_in);
+        return -1;
+    }
+    LIST_INIT (&hw->sw_head);
+    hw->active = 1;
+    hw->samples = hw->bufsize >> hw->info.shift;
+    hw->conv =
+        mixeng_conv
+        [nchannels == 2]
+        [hw->info.sign]
+        [hw->info.swap_endian]
+        [hw->info.bits == 16];
+    if (audio_pcm_hw_alloc_resources_in (hw)) {
+        audio_pcm_hw_free_resources_in (hw);
+        return -1;
+    }
+    return 0;
 }
 
-int pcm_hw_get_live (HWVoice *hw)
+static uint64_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
 {
-    int i, alive = 0, live = hw->samples;
+    SWVoiceIn *sw;
+    int m = hw->total_samples_captured;
 
-    for (i = 0; i < hw->nb_voices; i++) {
-        if (hw->pvoice[i]->live) {
-            live = audio_MIN (hw->pvoice[i]->live, live);
-            alive += 1;
+    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;
+}
 
-    if (alive)
-        return live;
-    else
-        return -1;
+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 pcm_hw_get_live2 (HWVoice *hw, int *nb_active)
+/*
+ * Soft voice (capture)
+ */
+static void audio_pcm_sw_free_resources_in (SWVoiceIn *sw)
 {
-    int i, alive = 0, live = hw->samples;
+    if (sw->conv_buf) {
+        qemu_free (sw->conv_buf);
+    }
 
-    *nb_active = 0;
-    for (i = 0; i < hw->nb_voices; i++) {
-        if (hw->pvoice[i]->live) {
-            if (hw->pvoice[i]->live < live) {
-                *nb_active = hw->pvoice[i]->active != 0;
-                live = hw->pvoice[i]->live;
-            }
-            alive += 1;
-        }
+    if (sw->rate) {
+        st_rate_stop (sw->rate);
     }
 
-    if (alive)
-        return live;
-    else
-        return -1;
+    sw->conv_buf = NULL;
+    sw->rate = NULL;
 }
 
-void pcm_hw_dec_live (HWVoice *hw, int decr)
+static int audio_pcm_sw_alloc_resources_in (SWVoiceIn *sw)
 {
-    int i;
+    int samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
+    sw->conv_buf = qemu_mallocz (samples * sizeof (st_sample_t));
+    if (!sw->conv_buf) {
+        return -1;
+    }
 
-    for (i = 0; i < hw->nb_voices; i++) {
-        if (hw->pvoice[i]->live) {
-            hw->pvoice[i]->live -= decr;
-        }
+    sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
+    if (!sw->rate) {
+        qemu_free (sw->conv_buf);
+        sw->conv_buf = NULL;
+        return -1;
     }
+    return 0;
 }
 
-void pcm_hw_clear (HWVoice *hw, void *buf, int len)
+static int audio_pcm_sw_init_in (SWVoiceIn *sw, HWVoiceIn *hw, const char *name,
+                           int freq, int nchannels, audfmt_e fmt)
 {
-    if (!len)
-        return;
+    audio_pcm_init_info (&sw->info, freq, nchannels, fmt,
+                         /* None of the cards emulated by QEMU are big-endian
+                            hence following shortcut */
+                         audio_need_to_swap_endian (0));
+    sw->hw = hw;
+    sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
 
-    switch (hw->fmt) {
-    case AUD_FMT_S16:
-    case AUD_FMT_S8:
-        memset (buf, len << hw->shift, 0x00);
-        break;
+    sw->clip =
+        mixeng_clip
+        [nchannels == 2]
+        [sw->info.sign]
+        [sw->info.swap_endian]
+        [sw->info.bits == 16];
 
-    case AUD_FMT_U8:
-        memset (buf, len << hw->shift, 0x80);
-        break;
+    sw->name = qemu_strdup (name);
+    audio_pcm_sw_free_resources_in (sw);
+    return audio_pcm_sw_alloc_resources_in (sw);
+}
 
-    case AUD_FMT_U16:
-        {
-            unsigned int i;
-            uint16_t *p = buf;
-            int shift = hw->nchannels - 1;
+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;
 
-            for (i = 0; i < len << shift; i++) {
-                p[i] = INT16_MAX;
-            }
-        }
-        break;
+    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 pcm_hw_write (SWVoice *sw, void *buf, int size)
+int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
 {
-    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
-    int ret = 0, pos = 0;
-    if (!sw)
-        return size;
+    HWVoiceIn *hw = sw->hw;
+    int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+    st_sample_t *src, *dst = sw->conv_buf;
 
-    hwsamples = sw->hw->samples;
-    samples = size >> sw->shift;
+    rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
 
-    if (!sw->live) {
-        sw->wpos = sw->hw->rpos;
+    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;
     }
-    wpos = sw->wpos;
-    live = sw->live;
-    dead = hwsamples - live;
-    swlim = (dead * ((int64_t) INT_MAX)) / sw->ratio;
-    swlim = audio_MIN (swlim, samples);
 
-    ldebug ("size=%d live=%d dead=%d swlim=%d wpos=%d\n",
-           size, live, dead, swlim, wpos);
-    if (swlim)
-        sw->conv (sw->buf, buf, swlim);
+    swlim = (live * sw->ratio) >> 32;
+    swlim = audio_MIN (swlim, samples);
 
     while (swlim) {
-        dead = hwsamples - live;
-        left = hwsamples - wpos;
-        blck = audio_MIN (dead, left);
-        if (!blck) {
-            /* dolog ("swlim=%d\n", swlim); */
+        src = hw->conv_buf + rpos;
+        isamp = hw->wpos - rpos;
+        /* XXX: <= ? */
+        if (isamp <= 0) {
+            isamp = hw->samples - rpos;
+        }
+
+        if (!isamp) {
             break;
         }
-        isamp = swlim;
-        osamp = blck;
-        st_rate_flow (sw->rate, sw->buf + pos, sw->hw->mix_buf + wpos, &isamp, &osamp);
-        ret += isamp;
-        swlim -= isamp;
-        pos += isamp;
-        live += osamp;
-        wpos = (wpos + osamp) % hwsamples;
+        osamp = swlim;
+
+        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
+            dolog ("osamp=%d\n", osamp);
+        }
+
+        st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
+        swlim -= osamp;
+        rpos = (rpos + isamp) % hw->samples;
+        dst += osamp;
+        ret += osamp;
+        total += isamp;
     }
 
-    sw->wpos = wpos;
-    sw->live = live;
-    return ret << sw->shift;
+    sw->clip (buf, sw->conv_buf, ret);
+    sw->total_hw_samples_acquired += total;
+    return ret << sw->info.shift;
 }
 
-int pcm_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+/*
+ * Hard voice (playback)
+ */
+static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
 {
-    int sign = 0, bits = 8;
+    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;
+}
 
-    pcm_hw_fini (hw);
-    ldebug ("pcm_hw_init: %d %d %d\n", freq, nchannels, fmt);
-    if (hw->pcm_ops->init (hw, freq, nchannels, fmt)) {
-        memset (hw, 0, audio_state.drv->voice_size);
+static void audio_pcm_hw_free_resources_out (HWVoiceOut *hw)
+{
+    if (hw->mix_buf) {
+        qemu_free (hw->mix_buf);
+    }
+
+    hw->mix_buf = NULL;
+}
+
+static int audio_pcm_hw_alloc_resources_out (HWVoiceOut *hw)
+{
+    hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
+    if (!hw->mix_buf) {
         return -1;
     }
 
-    switch (hw->fmt) {
-    case AUD_FMT_S8:
-        sign = 1;
-    case AUD_FMT_U8:
-        break;
+    return 0;
+}
 
-    case AUD_FMT_S16:
-        sign = 1;
-    case AUD_FMT_U16:
-        bits = 16;
-        break;
+static int audio_pcm_hw_init_out (HWVoiceOut *hw, int freq,
+                            int nchannels, audfmt_e fmt)
+{
+    audio_pcm_hw_fini_out (hw);
+    if (hw->pcm_ops->init_out (hw, freq, nchannels, fmt)) {
+        memset (hw, 0, audio_state.drv->voice_size_out);
+        return -1;
     }
 
-    hw->nb_voices = 0;
+    LIST_INIT (&hw->sw_head);
     hw->active = 1;
-    hw->shift = (hw->nchannels == 2) + (bits == 16);
-    hw->bytes_per_second = hw->freq << hw->shift;
-    hw->align = (1 << hw->shift) - 1;
-    hw->samples = hw->bufsize >> hw->shift;
-    hw->clip = mixeng_clip[hw->nchannels == 2][sign][bits == 16];
-    if (pcm_hw_alloc_resources (hw)) {
-        pcm_hw_fini (hw);
+    hw->samples = hw->bufsize >> hw->info.shift;
+    hw->clip =
+        mixeng_clip
+        [nchannels == 2]
+        [hw->info.sign]
+        [hw->info.swap_endian]
+        [hw->info.bits == 16];
+    if (audio_pcm_hw_alloc_resources_out (hw)) {
+        audio_pcm_hw_fini_out (hw);
         return -1;
     }
     return 0;
 }
 
-static int dist (void *hw)
+int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
 {
-    if (hw) {
-        return (((uint8_t *) hw - (uint8_t *) hw_voices)
-                / audio_state.drv->voice_size) + 1;
+    int smin;
+
+    smin = audio_pcm_hw_find_min_out (hw, nb_live);
+
+    if (!*nb_live) {
+        return 0;
     }
     else {
-        return 0;
+        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;
     }
 }
 
-#define ADVANCE(hw) \
-    ((hw) ? advance (hw, audio_state.drv->voice_size) : hw_voices)
-
-HWVoice *pcm_hw_find_any (HWVoice *hw)
+int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
 {
-    int i = dist (hw);
-    for (; i < audio_state.nb_hw_voices; i++) {
-        hw = ADVANCE (hw);
-        return hw;
+    int nb_live;
+    int live;
+
+    live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
+    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+        return 0;
     }
-    return NULL;
+    return live;
 }
 
-HWVoice *pcm_hw_find_any_active (HWVoice *hw)
+/*
+ * Soft voice (playback)
+ */
+static void audio_pcm_sw_free_resources_out (SWVoiceOut *sw)
 {
-    int i = dist (hw);
-    for (; i < audio_state.nb_hw_voices; i++) {
-        hw = ADVANCE (hw);
-        if (hw->active)
-            return hw;
+    if (sw->buf) {
+        qemu_free (sw->buf);
     }
-    return NULL;
-}
 
-HWVoice *pcm_hw_find_any_active_enabled (HWVoice *hw)
-{
-    int i = dist (hw);
-    for (; i < audio_state.nb_hw_voices; i++) {
-        hw = ADVANCE (hw);
-        if (hw->active && hw->enabled)
-            return hw;
+    if (sw->rate) {
+        st_rate_stop (sw->rate);
     }
-    return NULL;
+
+    sw->buf = NULL;
+    sw->rate = NULL;
 }
 
-HWVoice *pcm_hw_find_any_passive (HWVoice *hw)
+static int audio_pcm_sw_alloc_resources_out (SWVoiceOut *sw)
 {
-    int i = dist (hw);
-    for (; i < audio_state.nb_hw_voices; i++) {
-        hw = ADVANCE (hw);
-        if (!hw->active)
-            return hw;
+    sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t));
+    if (!sw->buf) {
+        return -1;
+    }
+
+    sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
+    if (!sw->rate) {
+        qemu_free (sw->buf);
+        sw->buf = NULL;
+        return -1;
     }
-    return NULL;
+    return 0;
 }
 
-HWVoice *pcm_hw_find_specific (HWVoice *hw, int freq,
-                               int nchannels, audfmt_e fmt)
+static int audio_pcm_sw_init_out (SWVoiceOut *sw, HWVoiceOut *hw,
+                            const char *name, int freq,
+                            int nchannels, audfmt_e fmt)
 {
-    while ((hw = pcm_hw_find_any_active (hw))) {
-        if (hw->freq == freq &&
-            hw->nchannels == nchannels &&
-            hw->fmt == fmt)
-            return hw;
-    }
-    return NULL;
+    audio_pcm_init_info (&sw->info, freq, nchannels, fmt,
+                         /* None of the cards emulated by QEMU are big-endian
+                            hence following shortcut */
+                         audio_need_to_swap_endian (0));
+    sw->hw = hw;
+    sw->empty = 1;
+    sw->active = 0;
+    sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
+    sw->total_hw_samples_mixed = 0;
+
+    sw->conv =
+        mixeng_conv
+        [nchannels == 2]
+        [sw->info.sign]
+        [sw->info.swap_endian]
+        [sw->info.bits == 16];
+    sw->name = qemu_strdup (name);
+
+    audio_pcm_sw_free_resources_out (sw);
+    return audio_pcm_sw_alloc_resources_out (sw);
 }
 
-HWVoice *pcm_hw_add (int freq, int nchannels, audfmt_e fmt)
+int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
 {
-    HWVoice *hw;
+    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+    int ret = 0, pos = 0, total = 0;
+
+    if (!sw) {
+        return size;
+    }
 
-    if (audio_state.fixed_format) {
-        freq = audio_state.fixed_freq;
-        nchannels = audio_state.fixed_channels;
-        fmt = audio_state.fixed_fmt;
+    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;
     }
 
-    hw = pcm_hw_find_specific (NULL, freq, nchannels, fmt);
+    if (live == hwsamples) {
+        return 0;
+    }
 
-    if (hw)
-        return hw;
+    wpos = (sw->hw->rpos + live) % hwsamples;
+    samples = size >> sw->info.shift;
 
-    hw = pcm_hw_find_any_passive (NULL);
-    if (hw) {
-        hw->pcm_ops = audio_state.drv->pcm_ops;
-        if (!hw->pcm_ops)
-            return NULL;
+    dead = hwsamples - live;
+    swlim = ((int64_t) dead << 32) / sw->ratio;
+    swlim = audio_MIN (swlim, samples);
+    if (swlim) {
+        sw->conv (sw->buf, buf, swlim, &sw->vol);
+    }
 
-        if (pcm_hw_init (hw, freq, nchannels, fmt)) {
-            pcm_hw_gc (hw);
-            return NULL;
+    while (swlim) {
+        dead = hwsamples - live;
+        left = hwsamples - wpos;
+        blck = audio_MIN (dead, left);
+        if (!blck) {
+            break;
         }
-        else
-            return hw;
+        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;
     }
 
-    return pcm_hw_find_any (NULL);
+    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, hw %d\n",
+        sw->name,
+        size >> sw->info.shift,
+        ret,
+        sw->total_hw_samples_mixed,
+        sw->hw->total_samples_played
+        );
+#endif
+
+    return ret << sw->info.shift;
 }
 
-int pcm_hw_add_sw (HWVoice *hw, SWVoice *sw)
+#ifdef DEBUG_AUDIO
+static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
 {
-    SWVoice **pvoice = qemu_mallocz ((hw->nb_voices + 1) * sizeof (sw));
-    if (!pvoice)
-        return -1;
-
-    memcpy (pvoice, hw->pvoice, hw->nb_voices * sizeof (sw));
-    qemu_free (hw->pvoice);
-    hw->pvoice = pvoice;
-    hw->pvoice[hw->nb_voices++] = sw;
-    return 0;
+    dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
+           cap, info->bits, info->sign, info->freq, info->nchannels);
 }
+#endif
 
-int pcm_hw_del_sw (HWVoice *hw, SWVoice *sw)
-{
-    int i, j;
-    if (hw->nb_voices > 1) {
-        SWVoice **pvoice = qemu_mallocz ((hw->nb_voices - 1) * sizeof (sw));
+#define DAC
+#include "audio_template.h"
+#undef DAC
+#include "audio_template.h"
 
-        if (!pvoice) {
-            dolog ("Can not maintain consistent state (not enough memory)\n");
-            return -1;
-        }
+int AUD_write (SWVoiceOut *sw, void *buf, int size)
+{
+    int bytes;
 
-        for (i = 0, j = 0; i < hw->nb_voices; i++) {
-            if (j >= hw->nb_voices - 1) {
-                dolog ("Can not maintain consistent state "
-                       "(invariant violated)\n");
-                return -1;
-            }
-            if (hw->pvoice[i] != sw)
-                pvoice[j++] = hw->pvoice[i];
-        }
-        qemu_free (hw->pvoice);
-        hw->pvoice = pvoice;
-        hw->nb_voices -= 1;
+    if (!sw) {
+        /* XXX: Consider options */
+        return size;
     }
-    else {
-        qemu_free (hw->pvoice);
-        hw->pvoice = NULL;
-        hw->nb_voices = 0;
+
+    if (!sw->hw->enabled) {
+        dolog ("Writing to disabled voice %s\n", sw->name);
+        return 0;
     }
-    return 0;
+
+    bytes = sw->hw->pcm_ops->write (sw, buf, size);
+    return bytes;
 }
 
-SWVoice *pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt)
+int AUD_read (SWVoiceIn *sw, void *buf, int size)
 {
-    SWVoice *sw;
-    HWVoice *hw;
-
-    sw = qemu_mallocz (sizeof (*sw));
-    if (!sw)
-        goto err1;
-
-    hw = pcm_hw_add (freq, nchannels, fmt);
-    if (!hw)
-        goto err2;
+    int bytes;
 
-    if (pcm_hw_add_sw (hw, sw))
-        goto err3;
+    if (!sw) {
+        /* XXX: Consider options */
+        return size;
+    }
 
-    if (pcm_sw_init (sw, hw, freq, nchannels, fmt))
-        goto err4;
+    if (!sw->hw->enabled) {
+        dolog ("Reading from disabled voice %s\n", sw->name);
+        return 0;
+    }
 
-    return sw;
+    bytes = sw->hw->pcm_ops->read (sw, buf, size);
+    return bytes;
+}
 
-err4:
-    pcm_hw_del_sw (hw, sw);
-err3:
-    pcm_hw_gc (hw);
-err2:
-    qemu_free (sw);
-err1:
-    return NULL;
+int AUD_get_buffer_size_out (SWVoiceOut *sw)
+{
+    return sw->hw->bufsize;
 }
 
-SWVoice *AUD_open (SWVoice *sw, const char *name,
-                   int freq, int nchannels, audfmt_e fmt)
+void AUD_set_active_out (SWVoiceOut *sw, int on)
 {
-    if (!audio_state.drv) {
-        return NULL;
-    }
+    HWVoiceOut *hw;
 
-    if (sw && freq == sw->freq && sw->nchannels == nchannels && sw->fmt == fmt) {
-        return sw;
+    if (!sw) {
+        return;
     }
 
-    if (sw) {
-        ldebug ("Different format %s %d %d %d\n",
-                name,
-                sw->freq == freq,
-                sw->nchannels == nchannels,
-                sw->fmt == fmt);
-    }
+    hw = sw->hw;
+    if (sw->active != on) {
+        SWVoiceOut *temp_sw;
 
-    if (nchannels != 1 && nchannels != 2) {
-        dolog ("Bogus channel count %d for voice %s\n", nchannels, name);
-        return NULL;
-    }
+        if (on) {
+            int total;
 
-    if (!audio_state.fixed_format && sw) {
-        pcm_sw_fini (sw);
-        pcm_hw_del_sw (sw->hw, sw);
-        pcm_hw_gc (sw->hw);
-        if (sw->name) {
-            qemu_free (sw->name);
-            sw->name = NULL;
-        }
-        qemu_free (sw);
-        sw = NULL;
-    }
+            hw->pending_disable = 0;
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
+            }
 
-    if (sw) {
-        HWVoice *hw = sw->hw;
-        if (!hw) {
-            dolog ("Internal logic error voice %s has no hardware store\n",
-                   name);
-            return sw;
+            if (sw->empty) {
+                total = 0;
+            }
         }
+        else {
+            if (hw->enabled) {
+                int nb_active = 0;
 
-        if (pcm_sw_init (sw, hw, freq, nchannels, fmt)) {
-            pcm_sw_fini (sw);
-            pcm_hw_del_sw (hw, sw);
-            pcm_hw_gc (hw);
-            if (sw->name) {
-                qemu_free (sw->name);
-                sw->name = NULL;
+                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;
             }
-            qemu_free (sw);
-            return NULL;
         }
+        sw->active = on;
     }
-    else {
-        sw = pcm_create_voice_pair (freq, nchannels, fmt);
-        if (!sw) {
-            dolog ("Failed to create voice %s\n", name);
-            return NULL;
-        }
+}
+
+void AUD_set_active_in (SWVoiceIn *sw, int on)
+{
+    HWVoiceIn *hw;
+
+    if (!sw) {
+        return;
     }
 
-    if (sw->name) {
-        qemu_free (sw->name);
-        sw->name = NULL;
+    hw = sw->hw;
+    if (sw->active != on) {
+        SWVoiceIn *temp_sw;
+
+        if (on) {
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
+            }
+            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;
     }
-    sw->name = qemu_strdup (name);
-    return sw;
 }
 
-void AUD_close (SWVoice *sw)
+static int audio_get_avail (SWVoiceIn *sw)
 {
-    if (!sw)
-        return;
+    int live;
+
+    if (!sw) {
+        return 0;
+    }
 
-    pcm_sw_fini (sw);
-    pcm_hw_del_sw (sw->hw, sw);
-    pcm_hw_gc (sw->hw);
-    if (sw->name) {
-        qemu_free (sw->name);
-        sw->name = NULL;
+    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;
     }
-    qemu_free (sw);
+
+    ldebug (
+        "%s: get_avail live %d ret %lld\n",
+        sw->name,
+        live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+        );
+
+    return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
 }
 
-int AUD_write (SWVoice *sw, void *buf, int size)
+static int audio_get_free (SWVoiceOut *sw)
 {
-    int bytes;
+    int live, dead;
 
-    if (!sw->hw->enabled)
-        dolog ("Writing to disabled voice %s\n", sw->name);
-    bytes = sw->hw->pcm_ops->write (sw, buf, size);
-    return bytes;
+    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);
+    }
+
+    dead = sw->hw->samples - live;
+
+#ifdef DEBUG_OUT
+    dolog ("%s: get_free live %d dead %d ret %lld\n",
+           sw->name,
+           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+#endif
+
+    return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
 }
 
-void AUD_run (void)
+static void audio_run_out (void)
 {
-    HWVoice *hw = NULL;
+    HWVoiceOut *hw = NULL;
+    SWVoiceOut *sw;
 
-    while ((hw = pcm_hw_find_any_active_enabled (hw))) {
-        int i;
-        if (hw->pending_disable && pcm_hw_get_live (hw) <= 0) {
+    while ((hw = audio_pcm_hw_find_any_active_enabled_out (hw))) {
+        int played;
+        int live, free, nb_live;
+
+        live = audio_pcm_hw_get_live_out2 (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);
+        }
+
+        if (hw->pending_disable && !nb_live) {
+#ifdef DEBUG_OUT
+            dolog ("Disabling voice\n");
+#endif
             hw->enabled = 0;
-            hw->pcm_ops->ctl (hw, VOICE_DISABLE);
-            for (i = 0; i < hw->nb_voices; i++) {
-                hw->pvoice[i]->live = 0;
-                /* hw->pvoice[i]->old_ticks = 0; */
+            hw->pending_disable = 0;
+            hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+            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;
         }
 
-        hw->pcm_ops->run (hw);
-        assert (hw->rpos < hw->samples);
-        for (i = 0; i < hw->nb_voices; i++) {
-            SWVoice *sw = hw->pvoice[i];
-            if (!sw->active && !sw->live && sw->old_ticks) {
-                int64_t delta = qemu_get_clock (vm_clock) - sw->old_ticks;
-                if (delta > audio_state.ticks_threshold) {
-                    ldebug ("resetting old_ticks for %s\n", sw->name);
-                    sw->old_ticks = 0;
+        played = hw->pcm_ops->run_out (hw);
+        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 total %d\n", played, hw->total_samples_played);
+#endif
+
+        if (played) {
+            hw->ts_helper += played;
+        }
+
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        again:
+            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;
+
+                if (!sw->active && !sw->callback.fn) {
+                    SWVoiceOut *temp = sw->entries.le_next;
+
+#ifdef DEBUG_PLIVE
+                    dolog ("Finishing with old voice\n");
+#endif
+                    AUD_close_out (sw);
+                    sw = temp;
+                    if (sw) {
+                        goto again;
+                    }
+                    else {
+                        break;
+                    }
+                }
+            }
+
+            if (sw->active) {
+                free = audio_get_free (sw);
+                if (free > 0) {
+                    sw->callback.fn (sw->callback.opaque, free);
                 }
             }
         }
     }
 }
 
-int AUD_get_free (SWVoice *sw)
+static void audio_run_in (void)
 {
-    int free;
+    HWVoiceIn *hw = NULL;
 
-    if (!sw)
-        return 4096;
+    while ((hw = audio_pcm_hw_find_any_active_enabled_in (hw))) {
+        SWVoiceIn *sw;
+        int captured, min;
 
-    free = ((sw->hw->samples - sw->live) << sw->hw->shift) * sw->ratio
-        / INT_MAX;
+        captured = hw->pcm_ops->run_in (hw);
 
-    free &= ~sw->hw->align;
-    if (!free) return 0;
+        min = audio_pcm_hw_find_min_in (hw);
+        hw->total_samples_captured += captured - min;
+        hw->ts_helper += captured;
 
-    return free;
-}
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            sw->total_hw_samples_acquired -= min;
 
-int AUD_get_buffer_size (SWVoice *sw)
-{
-    return sw->hw->bufsize;
-}
+            if (sw->active) {
+                int avail;
 
-void AUD_adjust (SWVoice *sw, int bytes)
-{
-    if (!sw)
-        return;
-    sw->old_ticks += (ticks_per_sec * (int64_t) bytes) / sw->bytes_per_second;
+                avail = audio_get_avail (sw);
+                if (avail > 0) {
+                    sw->callback.fn (sw->callback.opaque, avail);
+                }
+            }
+        }
+    }
 }
 
-void AUD_reset (SWVoice *sw)
-{
-    sw->active = 0;
-    sw->old_ticks = 0;
-}
+static struct audio_option audio_options[] = {
+    /* DAC */
+    {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &audio_state.fixed_settings_out,
+     "Use fixed settings for host DAC", NULL, 0},
 
-int AUD_calc_elapsed (SWVoice *sw)
-{
-    int64_t now, delta, bytes;
-    int dead, swlim;
+    {"DAC_FIXED_FREQ", AUD_OPT_INT, &audio_state.fixed_freq_out,
+     "Frequency for fixed host DAC", NULL, 0},
 
-    if (!sw)
-        return 0;
+    {"DAC_FIXED_FMT", AUD_OPT_FMT, &audio_state.fixed_fmt_out,
+     "Format for fixed host DAC", NULL, 0},
 
-    now = qemu_get_clock (vm_clock);
-    delta = now - sw->old_ticks;
-    bytes = (delta * sw->bytes_per_second) / ticks_per_sec;
-    if (delta < 0) {
-        dolog ("whoops delta(<0)=%lld\n", delta);
-        return 0;
-    }
+    {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &audio_state.fixed_channels_out,
+     "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},
 
-    dead = sw->hw->samples - sw->live;
-    swlim = ((dead * (int64_t) INT_MAX) / sw->ratio);
+    {"DAC_VOICES", AUD_OPT_INT, &audio_state.nb_hw_voices_out,
+     "Number of voices for DAC", NULL, 0},
 
-    if (bytes > swlim) {
-        return swlim;
-    }
-    else {
-        return bytes;
-    }
-}
+    /* ADC */
+    {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &audio_state.fixed_settings_out,
+     "Use fixed settings for host ADC", NULL, 0},
 
-void AUD_enable (SWVoice *sw, int on)
-{
-    int i;
-    HWVoice *hw;
+    {"ADC_FIXED_FREQ", AUD_OPT_INT, &audio_state.fixed_freq_out,
+     "Frequency for fixed ADC", NULL, 0},
 
-    if (!sw)
-        return;
+    {"ADC_FIXED_FMT", AUD_OPT_FMT, &audio_state.fixed_fmt_out,
+     "Format for fixed ADC", NULL, 0},
 
-    hw = sw->hw;
-    if (on) {
-        if (!sw->live)
-            sw->wpos = sw->hw->rpos;
-        if (!sw->old_ticks) {
-            sw->old_ticks = qemu_get_clock (vm_clock);
+    {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &audio_state.fixed_channels_in,
+     "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},
+
+    {"ADC_VOICES", AUD_OPT_INT, &audio_state.nb_hw_voices_out,
+     "Number of voices for ADC", NULL, 0},
+
+    /* Misc */
+    {"TIMER_PERIOD", AUD_OPT_INT, &audio_state.period.usec,
+     "Timer period in microseconds (0 - try lowest possible)", NULL, 0},
+
+    {"PLIVE", AUD_OPT_BOOL, &audio_state.plive,
+     "(undocumented)", NULL, 0},
+
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+void AUD_help (void)
+{
+    size_t i;
+
+    audio_process_options ("AUDIO", audio_options);
+    for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+        struct audio_driver *d = drvtab[i];
+        if (d->options) {
+            audio_process_options (d->name, d->options);
         }
     }
 
-    if (sw->active != on) {
-        if (on) {
-            hw->pending_disable = 0;
-            if (!hw->enabled) {
-                hw->enabled = 1;
-                for (i = 0; i < hw->nb_voices; i++) {
-                    ldebug ("resetting voice\n");
-                    sw = hw->pvoice[i];
-                    sw->old_ticks = qemu_get_clock (vm_clock);
-                }
-                hw->pcm_ops->ctl (hw, VOICE_ENABLE);
-            }
+    printf ("Audio options:\n");
+    audio_print_options ("AUDIO", audio_options);
+    printf ("\n");
+
+    printf ("Available drivers:\n");
+
+    for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+        struct audio_driver *d = drvtab[i];
+
+        printf ("Name: %s\n", d->name);
+        printf ("Description: %s\n", d->descr);
+
+        switch (d->max_voices_out) {
+        case 0:
+            printf ("Does not support DAC\n");
+            break;
+        case 1:
+            printf ("One DAC voice\n");
+            break;
+        case INT_MAX:
+            printf ("Theoretically supports many DAC voices\n");
+            break;
+        default:
+            printf ("Theoretically supports upto %d DAC voices\n",
+                     d->max_voices_out);
+            break;
         }
-        else {
-            if (hw->enabled && !hw->pending_disable) {
-                int nb_active = 0;
-                for (i = 0; i < hw->nb_voices; i++) {
-                    nb_active += hw->pvoice[i]->active != 0;
-                }
 
-                if (nb_active == 1) {
-                    hw->pending_disable = 1;
-                }
-            }
+        switch (d->max_voices_in) {
+        case 0:
+            printf ("Does not support ADC\n");
+            break;
+        case 1:
+            printf ("One ADC voice\n");
+            break;
+        case INT_MAX:
+            printf ("Theoretically supports many ADC voices\n");
+            break;
+        default:
+            printf ("Theoretically supports upto %d ADC voices\n",
+                     d->max_voices_in);
+            break;
         }
-        sw->active = on;
+
+        if (d->options) {
+            printf ("Options:\n");
+            audio_print_options (d->name, d->options);
+        }
+        else {
+            printf ("No options\n");
+        }
+        printf ("\n");
     }
-}
 
-static struct audio_output_driver *drvtab[] = {
-#ifdef CONFIG_OSS
-    &oss_output_driver,
-#endif
-#ifdef CONFIG_FMOD
-    &fmod_output_driver,
-#endif
-#ifdef CONFIG_SDL
-    &sdl_output_driver,
-#endif
-    &no_output_driver,
-#ifdef USE_WAV_AUDIO
-    &wav_output_driver,
+    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"
+        );
+}
+
+void audio_timer (void *opaque)
+{
+    AudioState *s = opaque;
 
-static int voice_init (struct audio_output_driver *drv)
+    audio_run_out ();
+    audio_run_in ();
+
+    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + s->period.ticks);
+}
+
+static int audio_driver_init (struct audio_driver *drv)
 {
+    if (drv->options) {
+        audio_process_options (drv->name, drv->options);
+    }
     audio_state.opaque = drv->init ();
+
     if (audio_state.opaque) {
-        if (audio_state.nb_hw_voices > drv->max_voices) {
-            dolog ("`%s' does not support %d multiple hardware channels\n"
-                   "Resetting to %d\n",
-                   drv->name, audio_state.nb_hw_voices, drv->max_voices);
-            audio_state.nb_hw_voices = drv->max_voices;
+        int i;
+        HWVoiceOut *hwo;
+        HWVoiceIn *hwi;
+
+        if (audio_state.nb_hw_voices_out > drv->max_voices_out) {
+            if (!drv->max_voices_out) {
+                dolog ("`%s' does not support DAC\n", drv->name);
+            }
+            else {
+                dolog (
+                    "`%s' does not support %d multiple DAC voicess\n"
+                    "Resetting to %d\n",
+                    drv->name,
+                    audio_state.nb_hw_voices_out,
+                    drv->max_voices_out
+                    );
+            }
+            audio_state.nb_hw_voices_out = drv->max_voices_out;
+        }
+
+        LIST_INIT (&hw_head_out);
+        hwo = qemu_mallocz (audio_state.nb_hw_voices_out * drv->voice_size_out);
+        if (!hwo) {
+            dolog (
+                "Not enough memory for %d `%s' DAC voices (each %d bytes)\n",
+                audio_state.nb_hw_voices_out,
+                drv->name,
+                drv->voice_size_out
+                );
+            drv->fini (audio_state.opaque);
+            return -1;
         }
-        hw_voices = qemu_mallocz (audio_state.nb_hw_voices * drv->voice_size);
-        if (hw_voices) {
-            audio_state.drv = drv;
-            return 1;
+
+        for (i = 0; i < audio_state.nb_hw_voices_out; ++i) {
+            LIST_INSERT_HEAD (&hw_head_out, hwo, entries);
+            hwo = advance (hwo, drv->voice_size_out);
         }
-        else {
-            dolog ("Not enough memory for %d `%s' voices (each %d bytes)\n",
-                   audio_state.nb_hw_voices, drv->name, drv->voice_size);
+
+        if (!drv->voice_size_in && drv->max_voices_in) {
+            ldebug ("warning: No ADC voice size defined for `%s'\n",
+                    drv->name);
+            drv->max_voices_in = 0;
+        }
+
+        if (!drv->voice_size_out && drv->max_voices_out) {
+            ldebug ("warning: No DAC voice size defined for `%s'\n",
+                    drv->name);
+        }
+
+        if (drv->voice_size_in && !drv->max_voices_in) {
+            ldebug ("warning: ADC voice size is %d for ADC less driver `%s'\n",
+                    drv->voice_size_out, drv->name);
+        }
+
+        if (drv->voice_size_out && !drv->max_voices_out) {
+            ldebug ("warning: DAC voice size is %d for DAC less driver `%s'\n",
+                    drv->voice_size_in, drv->name);
+        }
+
+        if (audio_state.nb_hw_voices_in > drv->max_voices_in) {
+            if (!drv->max_voices_in) {
+                ldebug ("`%s' does not support ADC\n", drv->name);
+            }
+            else {
+                dolog (
+                    "`%s' does not support %d multiple ADC voices\n"
+                    "Resetting to %d\n",
+                    drv->name,
+                    audio_state.nb_hw_voices_in,
+                    drv->max_voices_in
+                    );
+            }
+            audio_state.nb_hw_voices_in = drv->max_voices_in;
+        }
+
+        LIST_INIT (&hw_head_in);
+        hwi = qemu_mallocz (audio_state.nb_hw_voices_in * drv->voice_size_in);
+        if (!hwi) {
+            dolog (
+                "Not enough memory for %d `%s' ADC voices (each %d bytes)\n",
+                audio_state.nb_hw_voices_in,
+                drv->name,
+                drv->voice_size_in
+                );
+            qemu_free (hwo);
             drv->fini (audio_state.opaque);
-            return 0;
+            return -1;
         }
+
+        for (i = 0; i < audio_state.nb_hw_voices_in; ++i) {
+            LIST_INSERT_HEAD (&hw_head_in, hwi, entries);
+            hwi = advance (hwi, drv->voice_size_in);
+        }
+
+        audio_state.drv = drv;
+        return 0;
     }
     else {
-        dolog ("Could not init `%s' audio\n", drv->name);
-        return 0;
+        dolog ("Could not init `%s' audio driver\n", drv->name);
+        return -1;
     }
 }
 
 static void audio_vm_stop_handler (void *opaque, int reason)
 {
-    HWVoice *hw = NULL;
+    HWVoiceOut *hwo = NULL;
+    HWVoiceIn *hwi = NULL;
+    int op = reason ? VOICE_ENABLE : VOICE_DISABLE;
+
+    (void) opaque;
+    while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+        if (!hwo->pcm_ops) {
+            continue;
+        }
+
+        if (hwo->enabled != reason) {
+            hwo->pcm_ops->ctl_out (hwo, op);
+        }
+    }
 
-    while ((hw = pcm_hw_find_any (hw))) {
-        if (!hw->pcm_ops)
+    while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+        if (!hwi->pcm_ops) {
             continue;
+        }
 
-        hw->pcm_ops->ctl (hw, reason ? VOICE_ENABLE : VOICE_DISABLE);
+        if (hwi->enabled != reason) {
+            hwi->pcm_ops->ctl_in (hwi, op);
+        }
     }
 }
 
 static void audio_atexit (void)
 {
-    HWVoice *hw = NULL;
+    HWVoiceOut *hwo = NULL;
+    HWVoiceIn *hwi = NULL;
+
+    while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+        if (!hwo->pcm_ops) {
+            continue;
+        }
+
+        if (hwo->enabled) {
+            hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
+        }
+        hwo->pcm_ops->fini_out (hwo);
+    }
 
-    while ((hw = pcm_hw_find_any (hw))) {
-        if (!hw->pcm_ops)
+    while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+        if (!hwi->pcm_ops) {
             continue;
+        }
 
-        hw->pcm_ops->ctl (hw, VOICE_DISABLE);
-        hw->pcm_ops->fini (hw);
+        if (hwi->enabled) {
+            hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
+        }
+        hwi->pcm_ops->fini_in (hwi);
     }
     audio_state.drv->fini (audio_state.opaque);
 }
 
 static void audio_save (QEMUFile *f, void *opaque)
 {
+    (void) f;
+    (void) opaque;
 }
 
 static int audio_load (QEMUFile *f, void *opaque, int version_id)
 {
-    if (version_id != 1)
+    (void) f;
+    (void) opaque;
+
+    if (version_id != 1) {
         return -EINVAL;
+    }
 
     return 0;
 }
 
 void AUD_init (void)
 {
-    int i;
+    size_t i;
     int done = 0;
     const char *drvname;
+    AudioState *s = &audio_state;
 
-    audio_state.fixed_format =
-        !!audio_get_conf_int (QC_FIXED_FORMAT, audio_state.fixed_format);
-    audio_state.fixed_freq =
-        audio_get_conf_int (QC_FIXED_FREQ, audio_state.fixed_freq);
-    audio_state.nb_hw_voices =
-        audio_get_conf_int (QC_VOICES, audio_state.nb_hw_voices);
+    audio_process_options ("AUDIO", audio_options);
 
-    if (audio_state.nb_hw_voices <= 0) {
-        dolog ("Bogus number of voices %d, resetting to 1\n",
-               audio_state.nb_hw_voices);
+    if (s->nb_hw_voices_out <= 0) {
+        dolog ("Bogus number of DAC voices %d\n",
+               s->nb_hw_voices_out);
+        s->nb_hw_voices_out = 1;
+    }
+
+    if (s->nb_hw_voices_in <= 0) {
+        dolog ("Bogus number of ADC voices %d\n",
+               s->nb_hw_voices_in);
+        s->nb_hw_voices_in = 1;
+    }
+
+    {
+        int def;
+        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
+    }
+
+    s->ts = qemu_new_timer (vm_clock, audio_timer, s);
+    if (!s->ts) {
+        dolog ("Can not create audio timer\n");
+        return;
     }
 
-    drvname = audio_get_conf_str (QC_AUDIO_DRV, NULL);
     if (drvname) {
         int found = 0;
+
         for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
             if (!strcmp (drvname, drvtab[i]->name)) {
-                done = voice_init (drvtab[i]);
+                done = !audio_driver_init (drvtab[i]);
                 found = 1;
                 break;
             }
         }
+
         if (!found) {
             dolog ("Unknown audio driver `%s'\n", drvname);
+            dolog ("Run with -audio-help to list available drivers\n");
         }
     }
 
@@ -895,17 +1624,32 @@ void AUD_init (void)
 
     if (!done) {
         for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
-            if (drvtab[i]->can_be_default)
-                done = voice_init (drvtab[i]);
+            if (drvtab[i]->can_be_default) {
+                done = !audio_driver_init (drvtab[i]);
+            }
         }
     }
 
-    audio_state.ticks_threshold = ticks_per_sec / 50;
-    audio_state.freq_threshold = 100;
-
     register_savevm ("audio", 0, 1, audio_save, audio_load, NULL);
     if (!done) {
-        dolog ("Can not initialize audio subsystem\n");
-        voice_init (&no_output_driver);
+        if (audio_driver_init (&no_audio_driver)) {
+            dolog ("Can not initialize audio subsystem\n");
+        }
+        else {
+            dolog ("warning: using timer based audio emulation\n");
+        }
     }
+
+    if (s->period.usec <= 0) {
+        if (s->period.usec < 0) {
+            dolog ("warning: timer period is negative - %d treating as zero\n",
+                   s->period.usec);
+        }
+        s->period.ticks = 1;
+    }
+    else {
+        s->period.ticks = (ticks_per_sec * s->period.usec) / 1000000;
+    }
+
+    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + s->period.ticks);
 }
diff --git a/audio/audio.h b/audio/audio.h
index 7520383a47..6dd2fd22e3 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -1,8 +1,8 @@
 /*
  * QEMU Audio subsystem header
- * 
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- * 
+ *
+ * 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
@@ -24,7 +24,7 @@
 #ifndef QEMU_AUDIO_H
 #define QEMU_AUDIO_H
 
-#include "mixeng.h"
+typedef void (*audio_callback_fn_t) (void *opaque, int avail);
 
 typedef enum {
   AUD_FMT_U8,
@@ -33,22 +33,60 @@ typedef enum {
   AUD_FMT_S16
 } audfmt_e;
 
-typedef struct SWVoice SWVoice;
+typedef struct SWVoiceOut SWVoiceOut;
+typedef struct SWVoiceIn SWVoiceIn;
+
+typedef struct QEMUAudioTimeStamp {
+    uint64_t old_ts;
+} QEMUAudioTimeStamp;
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap);
+void AUD_log (const char *cap, const char *fmt, ...)
+#ifdef __GNUC__
+    __attribute__ ((__format__ (__printf__, 2, 3)))
+#endif
+    ;
 
-SWVoice * AUD_open (SWVoice *sw, const char *name, int freq,
-                    int nchannels, audfmt_e fmt);
-void   AUD_init (void);
-void   AUD_log (const char *cap, const char *fmt, ...)
-    __attribute__ ((__format__ (__printf__, 2, 3)));;
-void   AUD_close (SWVoice *sw);
-int    AUD_write (SWVoice *sw, void *pcm_buf, int size);
-void   AUD_adjust (SWVoice *sw, int leftover);
-void   AUD_reset (SWVoice *sw);
-int    AUD_get_free (SWVoice *sw);
-int    AUD_get_buffer_size (SWVoice *sw);
-void   AUD_run (void);
-void   AUD_enable (SWVoice *sw, int on);
-int    AUD_calc_elapsed (SWVoice *sw);
+void AUD_init (void);
+void AUD_help (void);
+
+SWVoiceOut  *AUD_open_out (
+    SWVoiceOut *sw,
+    const char *name,
+    void *callback_opaque,
+    audio_callback_fn_t callback_fn,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    );
+void         AUD_close_out (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_time_stamp_get_elapsed_usec_out (SWVoiceOut *sw,
+                                                  QEMUAudioTimeStamp *ts);
+
+SWVoiceIn   *AUD_open_in (
+    SWVoiceIn *sw,
+    const char *name,
+    void *callback_opaque,
+    audio_callback_fn_t callback_fn,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    );
+void         AUD_close_in (SWVoiceIn *sw);
+int          AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+void         AUD_adjust_in (SWVoiceIn *sw, int leftover);
+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_time_stamp_get_elapsed_usec_in (SWVoiceIn *sw,
+                                                 QEMUAudioTimeStamp *ts);
 
 static inline void *advance (void *p, int incr)
 {
@@ -59,7 +97,21 @@ static inline void *advance (void *p, int incr)
 uint32_t popcount (uint32_t u);
 inline uint32_t lsbindex (uint32_t u);
 
+#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
 
 #endif  /* audio.h */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 0be2a61662..9d288292a7 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -1,8 +1,8 @@
 /*
  * QEMU Audio subsystem header
- * 
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- * 
+ *
+ * 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
@@ -24,140 +24,266 @@
 #ifndef QEMU_AUDIO_INT_H
 #define QEMU_AUDIO_INT_H
 
-#include "vl.h"
+#include "sys-queue.h"
+
+#ifdef CONFIG_COREAUDIO
+#define FLOAT_MIXENG
+/* #define RECIPROCAL */
+#endif
+#include "mixeng.h"
+
+int audio_bug (const char *funcname, int cond);
+
+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 *overridenp;
+    int overriden;
+};
+
+struct audio_callback {
+    void *opaque;
+    audio_callback_fn_t fn;
+};
 
-struct pcm_ops;
+struct audio_pcm_info {
+    int bits;
+    int sign;
+    int freq;
+    int nchannels;
+    int align;
+    int shift;
+    int bytes_per_second;
+    int swap_endian;
+};
 
-typedef struct HWVoice {
+typedef struct HWVoiceOut {
     int active;
     int enabled;
     int pending_disable;
     int valid;
-    int freq;
+    struct audio_pcm_info info;
 
     f_sample *clip;
-    audfmt_e fmt;
-    int nchannels;
-
-    int align;
-    int shift;
 
     int rpos;
     int bufsize;
+    uint64_t ts_helper;
 
-    int bytes_per_second;
     st_sample_t *mix_buf;
 
     int samples;
-    int64_t old_ticks;
-    int nb_voices;
-    struct SWVoice **pvoice;
-    struct pcm_ops *pcm_ops;
-} HWVoice;
+    LIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
+    struct audio_pcm_ops *pcm_ops;
+    LIST_ENTRY (HWVoiceOut) entries;
+} HWVoiceOut;
 
-extern struct pcm_ops no_pcm_ops;
-extern struct audio_output_driver no_output_driver;
+typedef struct HWVoiceIn {
+    int enabled;
+    int active;
+    struct audio_pcm_info info;
+
+    t_sample *conv;
 
-extern struct pcm_ops oss_pcm_ops;
-extern struct audio_output_driver oss_output_driver;
+    int wpos;
+    int bufsize;
+    int total_samples_captured;
+    uint64_t ts_helper;
 
-extern struct pcm_ops sdl_pcm_ops;
-extern struct audio_output_driver sdl_output_driver;
+    st_sample_t *conv_buf;
 
-extern struct pcm_ops wav_pcm_ops;
-extern struct audio_output_driver wav_output_driver;
+    int samples;
+    LIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
+    struct audio_pcm_ops *pcm_ops;
+    LIST_ENTRY (HWVoiceIn) entries;
+} HWVoiceIn;
 
-extern struct pcm_ops fmod_pcm_ops;
-extern struct audio_output_driver fmod_output_driver;
+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 fmod_audio_driver;
+extern struct audio_driver alsa_audio_driver;
+extern struct audio_driver coreaudio_audio_driver;
+extern struct audio_driver dsound_audio_driver;
+extern volume_t nominal_volume;
 
-struct audio_output_driver {
+struct audio_driver {
     const char *name;
+    const char *descr;
+    struct audio_option *options;
     void *(*init) (void);
     void (*fini) (void *);
-    struct pcm_ops *pcm_ops;
+    struct audio_pcm_ops *pcm_ops;
     int can_be_default;
-    int max_voices;
-    int voice_size;
+    int max_voices_out;
+    int max_voices_in;
+    int voice_size_out;
+    int voice_size_in;
 };
 
 typedef struct AudioState {
-    int fixed_format;
-    int fixed_freq;
-    int fixed_channels;
-    int fixed_fmt;
-    int nb_hw_voices;
-    int64_t ticks_threshold;
-    int freq_threshold;
+    int fixed_settings_out;
+    int fixed_freq_out;
+    int fixed_channels_out;
+    int fixed_fmt_out;
+    int nb_hw_voices_out;
+    int greedy_out;
+
+    int fixed_settings_in;
+    int fixed_freq_in;
+    int fixed_channels_in;
+    int fixed_fmt_in;
+    int nb_hw_voices_in;
+    int greedy_in;
+
     void *opaque;
-    struct audio_output_driver *drv;
-} AudioState;
-extern AudioState audio_state;
+    struct audio_driver *drv;
 
-struct SWVoice {
-    int freq;
-    audfmt_e fmt;
-    int nchannels;
+    QEMUTimer *ts;
+    union {
+        int usec;
+        int64_t ticks;
+    } period;
 
-    int shift;
-    int align;
+    int plive;
+} AudioState;
+extern AudioState audio_state;
 
+struct SWVoiceOut {
+    struct audio_pcm_info info;
     t_sample *conv;
-
-    int left;
-    int pos;
-    int bytes_per_second;
     int64_t ratio;
     st_sample_t *buf;
     void *rate;
+    int total_hw_samples_mixed;
+    int active;
+    int empty;
+    HWVoiceOut *hw;
+    char *name;
+    volume_t vol;
+    struct audio_callback callback;
+    LIST_ENTRY (SWVoiceOut) entries;
+};
 
-    int wpos;
-    int live;
+struct SWVoiceIn {
     int active;
-    int64_t old_ticks;
-    HWVoice *hw;
+    struct audio_pcm_info info;
+    int64_t ratio;
+    void *rate;
+    int total_hw_samples_acquired;
+    st_sample_t *conv_buf;
+    f_sample *clip;
+    HWVoiceIn *hw;
     char *name;
+    volume_t vol;
+    struct audio_callback callback;
+    LIST_ENTRY (SWVoiceIn) entries;
 };
 
-struct pcm_ops {
-    int  (*init)  (HWVoice *hw, int freq, int nchannels, audfmt_e fmt);
-    void (*fini)  (HWVoice *hw);
-    void (*run)   (HWVoice *hw);
-    int  (*write) (SWVoice *sw, void *buf, int size);
-    int  (*ctl)   (HWVoice *hw, int cmd, ...);
+struct audio_pcm_ops {
+    int  (*init_out)(HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt);
+    void (*fini_out)(HWVoiceOut *hw);
+    int  (*run_out) (HWVoiceOut *hw);
+    int  (*write)   (SWVoiceOut *sw, void *buf, int size);
+    int  (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+
+    int  (*init_in) (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt);
+    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, ...);
 };
 
-void      pcm_sw_free_resources (SWVoice *sw);
-int       pcm_sw_alloc_resources (SWVoice *sw);
-void      pcm_sw_fini (SWVoice *sw);
-int       pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
-                       int nchannels, audfmt_e fmt);
-
-void      pcm_hw_clear (HWVoice *hw, void *buf, int len);
-HWVoice * pcm_hw_find_any (HWVoice *hw);
-HWVoice * pcm_hw_find_any_active (HWVoice *hw);
-HWVoice * pcm_hw_find_any_passive (HWVoice *hw);
-HWVoice * pcm_hw_find_specific (HWVoice *hw, int freq,
-                                int nchannels, audfmt_e fmt);
-HWVoice * pcm_hw_add (int freq, int nchannels, audfmt_e fmt);
-int       pcm_hw_add_sw (HWVoice *hw, SWVoice *sw);
-int       pcm_hw_del_sw (HWVoice *hw, SWVoice *sw);
-SWVoice * pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt);
-
-void      pcm_hw_free_resources (HWVoice *hw);
-int       pcm_hw_alloc_resources (HWVoice *hw);
-void      pcm_hw_fini (HWVoice *hw);
-void      pcm_hw_gc (HWVoice *hw);
-int       pcm_hw_get_live (HWVoice *hw);
-int       pcm_hw_get_live2 (HWVoice *hw, int *nb_active);
-void      pcm_hw_dec_live (HWVoice *hw, int decr);
-int       pcm_hw_write (SWVoice *sw, void *buf, int len);
-
-int         audio_get_conf_int (const char *key, int defval);
-const char *audio_get_conf_str (const char *key, const char *defval);
-
-struct audio_output_driver;
+void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
+                          int nchannels, audfmt_e fmt, int swap_endian);
+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_get_live_out (HWVoiceOut *hw);
+int  audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live);
 
 #define VOICE_ENABLE 1
 #define VOICE_DISABLE 2
 
+static inline int audio_ring_dist (int dst, int src, int len)
+{
+    return (dst >= src) ? (dst - src) : (len - src + dst);
+}
+
+static inline int audio_need_to_swap_endian (int endianness)
+{
+#ifdef WORDS_BIGENDIAN
+    return endianness != 1;
+#else
+    return endianness != 0;
+#endif
+}
+
+#if defined __GNUC__
+#define GCC_ATTR __attribute__ ((__unused__, __format__ (__printf__, 1, 2)))
+#define INIT_FIELD(f) . f
+#define GCC_FMT_ATTR(n, m) __attribute__ ((__format__ (printf, n, m)))
+#else
+#define GCC_ATTR /**/
+#define INIT_FIELD(f) /**/
+#define GCC_FMT_ATTR(n, m)
+#endif
+
+static void GCC_ATTR dolog (const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+}
+
+#ifdef DEBUG
+static void GCC_ATTR ldebug (const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+}
+#else
+#if defined NDEBUG && defined __GNUC__
+#define ldebug(...)
+#elif defined NDEBUG && defined _MSC_VER
+#define ldebug __noop
+#else
+static void GCC_ATTR ldebug (const char *fmt, ...)
+{
+    (void) fmt;
+}
+#endif
+#endif
+
+#undef GCC_ATTR
+
+#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/audio/audio_template.h b/audio/audio_template.h
new file mode 100644
index 0000000000..25ea72fd41
--- /dev/null
+++ b/audio/audio_template.h
@@ -0,0 +1,401 @@
+/*
+ * 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 TYPE out
+#define HW glue (HWVoice, Out)
+#define SW glue (SWVoice, Out)
+#else
+#define TYPE in
+#define HW glue (HWVoice, In)
+#define SW glue (SWVoice, In)
+#endif
+
+static void glue (audio_pcm_sw_fini_, TYPE) (SW *sw)
+{
+    glue (audio_pcm_sw_free_resources_, TYPE) (sw);
+    if (sw->name) {
+        qemu_free (sw->name);
+        sw->name = NULL;
+    }
+}
+
+static void glue (audio_pcm_hw_add_sw_, TYPE) (HW *hw, SW *sw)
+{
+    LIST_INSERT_HEAD (&hw->sw_head, sw, entries);
+}
+
+static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
+{
+    LIST_REMOVE (sw, entries);
+}
+
+static void glue (audio_pcm_hw_fini_, TYPE) (HW *hw)
+{
+    if (hw->active) {
+        glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
+        glue (hw->pcm_ops->fini_, TYPE) (hw);
+        memset (hw, 0, glue (audio_state.drv->voice_size_, TYPE));
+    }
+}
+
+static void glue (audio_pcm_hw_gc_, TYPE) (HW *hw)
+{
+    if (!hw->sw_head.lh_first) {
+        glue (audio_pcm_hw_fini_, TYPE) (hw);
+    }
+}
+
+static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+{
+    return hw ? hw->entries.le_next : glue (hw_head_, TYPE).lh_first;
+}
+
+static HW *glue (audio_pcm_hw_find_any_active_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (hw->active) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_any_active_enabled_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (hw->active && hw->enabled) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_any_passive_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (!hw->active) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
+    HW *hw,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    while ((hw = glue (audio_pcm_hw_find_any_active_, TYPE) (hw))) {
+        if (audio_pcm_info_eq (&hw->info, freq, nchannels, fmt)) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_new_, TYPE) (
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    HW *hw;
+
+    hw = glue (audio_pcm_hw_find_any_passive_, TYPE) (NULL);
+    if (hw) {
+        hw->pcm_ops = audio_state.drv->pcm_ops;
+        if (!hw->pcm_ops) {
+            return NULL;
+        }
+
+        if (glue (audio_pcm_hw_init_, TYPE) (hw, freq, nchannels, fmt)) {
+            glue (audio_pcm_hw_gc_, TYPE) (hw);
+            return NULL;
+        }
+        else {
+            return hw;
+        }
+    }
+
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_, TYPE) (
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    HW *hw;
+
+    if (glue (audio_state.greedy_, TYPE)) {
+        hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+        if (hw) {
+            return hw;
+        }
+    }
+
+    hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, freq, nchannels, fmt);
+    if (hw) {
+        return hw;
+    }
+
+    hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+    if (hw) {
+        return hw;
+    }
+
+    return glue (audio_pcm_hw_find_any_active_, TYPE) (NULL);
+}
+
+static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
+    const char *name,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    SW *sw;
+    HW *hw;
+    int hw_freq = freq;
+    int hw_nchannels = nchannels;
+    int hw_fmt = fmt;
+
+    if (glue (audio_state.fixed_settings_, TYPE)) {
+        hw_freq = glue (audio_state.fixed_freq_, TYPE);
+        hw_nchannels = glue (audio_state.fixed_channels_, TYPE);
+        hw_fmt = glue (audio_state.fixed_fmt_, TYPE);
+    }
+
+    sw = qemu_mallocz (sizeof (*sw));
+    if (!sw) {
+        goto err1;
+    }
+
+    hw = glue (audio_pcm_hw_add_, TYPE) (hw_freq, hw_nchannels, hw_fmt);
+    if (!hw) {
+        goto err2;
+    }
+
+    glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
+
+    if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, freq, nchannels, fmt)) {
+        goto err3;
+    }
+
+    return sw;
+
+err3:
+    glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+    glue (audio_pcm_hw_gc_, TYPE) (hw);
+err2:
+    qemu_free (sw);
+err1:
+    return NULL;
+}
+
+void glue (AUD_close_, TYPE) (SW *sw)
+{
+    if (sw) {
+        glue (audio_pcm_sw_fini_, TYPE) (sw);
+        glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+        glue (audio_pcm_hw_gc_, TYPE) (sw->hw);
+        qemu_free (sw);
+    }
+}
+
+SW *glue (AUD_open_, TYPE) (
+    SW *sw,
+    const char *name,
+    void *callback_opaque ,
+    audio_callback_fn_t callback_fn,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+#ifdef DAC
+    int live = 0;
+    SW *old_sw = NULL;
+#endif
+
+    if (!callback_fn) {
+        dolog ("No callback specifed for voice `%s'\n", name);
+        goto fail;
+    }
+
+    if (nchannels != 1 && nchannels != 2) {
+        dolog ("Bogus channel count %d for voice `%s'\n", nchannels, name);
+        goto fail;
+    }
+
+    if (!audio_state.drv) {
+        dolog ("No audio driver defined\n");
+        goto fail;
+    }
+
+    if (sw && audio_pcm_info_eq (&sw->info, freq, nchannels, fmt)) {
+        return sw;
+    }
+
+#ifdef DAC
+    if (audio_state.plive && sw && (!sw->active && !sw->empty)) {
+        live = sw->total_hw_samples_mixed;
+
+#ifdef DEBUG_PLIVE
+        dolog ("Replacing voice %s with %d live samples\n", sw->name, live);
+        dolog ("Old %s freq %d, bits %d, channels %d\n",
+               sw->name, sw->info.freq, sw->info.bits, sw->info.nchannels);
+        dolog ("New %s freq %d, bits %d, channels %d\n",
+               name, freq, (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) ? 16 : 8,
+               nchannels);
+#endif
+
+        if (live) {
+            old_sw = sw;
+            old_sw->callback.fn = NULL;
+            sw = NULL;
+        }
+    }
+#endif
+
+    if (!glue (audio_state.fixed_settings_, TYPE) && sw) {
+        glue (AUD_close_, TYPE) (sw);
+        sw = NULL;
+    }
+
+    if (sw) {
+        HW *hw = sw->hw;
+
+        if (!hw) {
+            dolog ("Internal logic error voice %s has no hardware store\n",
+                   name);
+            goto fail;
+        }
+
+        if (glue (audio_pcm_sw_init_, TYPE) (
+                sw,
+                hw,
+                name,
+                freq,
+                nchannels,
+                fmt
+                )) {
+            goto fail;
+        }
+    }
+    else {
+        sw = glue (audio_pcm_create_voice_pair_, TYPE) (
+            name,
+            freq,
+            nchannels,
+            fmt);
+        if (!sw) {
+            dolog ("Failed to create voice %s\n", name);
+            goto fail;
+        }
+    }
+
+    if (sw) {
+        sw->vol = nominal_volume;
+        sw->callback.fn = callback_fn;
+        sw->callback.opaque = callback_opaque;
+
+#ifdef DAC
+        if (live) {
+            int mixed =
+                (live << old_sw->info.shift)
+                * old_sw->info.bytes_per_second
+                / sw->info.bytes_per_second;
+
+#ifdef DEBUG_PLIVE
+            dolog ("Silence will be mixed %d\n", mixed);
+#endif
+            sw->total_hw_samples_mixed += mixed;
+        }
+#endif
+
+#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) (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_time_stamp_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 %lld old %lld\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 (delta * sw->hw->info.freq) / 1000000;
+}
+
+#undef TYPE
+#undef HW
+#undef SW
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
new file mode 100644
index 0000000000..eee12386ce
--- /dev/null
+++ b/audio/coreaudio.c
@@ -0,0 +1,513 @@
+/*
+ * 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 "vl.h"
+
+#define AUDIO_CAP "coreaudio"
+#include "audio_int.h"
+
+#define DEVICE_BUFFER_FRAMES  (512)
+
+struct {
+    int buffer_frames;
+} conf = {
+    .buffer_frames = 512
+};
+
+typedef struct coreaudioVoiceOut {
+    HWVoiceOut hw;
+    pthread_mutex_t mutex;
+    AudioDeviceID outputDeviceID;
+    UInt32 audioDevicePropertyBufferSize;
+    AudioStreamBasicDescription outputStreamBasicDescription;
+    int isPlaying;
+    int live;
+    int decr;
+    int rpos;
+} coreaudioVoiceOut;
+
+static void coreaudio_logstatus (OSStatus status)
+{
+    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 %ld\n", 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, "Can not initialize %s\n", typ);
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    coreaudio_logstatus (status);
+}
+
+static int coreaudio_lock (coreaudioVoiceOut *core, const char *fn_name)
+{
+    int err;
+
+    err = pthread_mutex_lock (&core->mutex);
+    if (err) {
+        dolog ("Can 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 ("Can 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, decr;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    if (coreaudio_lock (core, "coreaudio_run_out")) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_out (hw);
+
+    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)
+{
+    unsigned int frame, frameCount;
+    float *out = outOutputData->mBuffers[0].mData;
+    HWVoiceOut *hw = hwptr;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
+    int rpos, live;
+    st_sample_t *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 = conf.buffer_frames;
+    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
+    }
+
+    /* cleanup */
+    mixeng_clear (src, frameCount);
+    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, int freq,
+                               int nchannels, audfmt_e fmt)
+{
+    OSStatus status;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+    UInt32 propertySize;
+    int err;
+    int bits = 8;
+    int endianess = 0;
+    const char *typ = "DAC";
+
+    /* create mutex */
+    err = pthread_mutex_init(&core->mutex, NULL);
+    if (err) {
+        dolog("Can not create mutex\nReason: %s\n", strerror (err));
+        return -1;
+    }
+
+    if (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) {
+        bits = 16;
+        endianess = 1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        freq,
+        nchannels,
+        fmt,
+        /* Following is irrelevant actually since we do not use
+           mixengs clipping routines */
+        audio_need_to_swap_endian (endianess)
+        );
+    hw->bufsize = 4 * conf.buffer_frames * nchannels * bits;
+
+    /* open default output device */
+    propertySize = sizeof(core->outputDeviceID);
+    status = AudioHardwareGetProperty(
+        kAudioHardwarePropertyDefaultOutputDevice,
+        &propertySize,
+        &core->outputDeviceID);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Can not get default output Device\n");
+        return -1;
+    }
+    if (core->outputDeviceID == kAudioDeviceUnknown) {
+        dolog ("Can not initialize %s - Unknown Audiodevice\n", typ);
+        return -1;
+    }
+
+    /* set Buffersize to conf.buffer_frames frames */
+    propertySize = sizeof(core->audioDevicePropertyBufferSize);
+    core->audioDevicePropertyBufferSize =
+        conf.buffer_frames * sizeof(float) * 2;
+    status = AudioDeviceSetProperty(
+        core->outputDeviceID,
+        NULL,
+        0,
+        false,
+        kAudioDevicePropertyBufferSize,
+        propertySize,
+        &core->audioDevicePropertyBufferSize);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Can not set device buffer size %d\n",
+                           kAudioDevicePropertyBufferSize);
+        return -1;
+    }
+
+    /* get Buffersize */
+    propertySize = sizeof(core->audioDevicePropertyBufferSize);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        false,
+        kAudioDevicePropertyBufferSize,
+        &propertySize,
+        &core->audioDevicePropertyBufferSize);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Can not get device buffer size\n");
+        return -1;
+    }
+
+    /* get StreamFormat */
+    propertySize = sizeof(core->outputStreamBasicDescription);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        false,
+        kAudioDevicePropertyStreamFormat,
+        &propertySize,
+        &core->outputStreamBasicDescription);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Can not get Device Stream properties\n");
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* set Samplerate */
+    core->outputStreamBasicDescription.mSampleRate = (Float64)freq;
+    propertySize = sizeof(core->outputStreamBasicDescription);
+    status = AudioDeviceSetProperty(
+        core->outputDeviceID,
+        0,
+        0,
+        0,
+        kAudioDevicePropertyStreamFormat,
+        propertySize,
+        &core->outputStreamBasicDescription);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Can not set samplerate %d\n", freq);
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* set Callback */
+    status = AudioDeviceAddIOProc(core->outputDeviceID, audioDeviceIOProc, hw);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Can not set IOProc\n");
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* start Playback */
+    if (!core->isPlaying) {
+        status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+        if (status != kAudioHardwareNoError) {
+            coreaudio_logerr2 (status, typ, "Can not start playback\n");
+            AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+            core->outputDeviceID = kAudioDeviceUnknown;
+            return -1;
+        }
+        core->isPlaying = 1;
+    }
+
+    return 0;
+}
+
+static void coreaudio_fini_out (HWVoiceOut *hw)
+{
+    OSStatus status;
+    int err;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    /* stop playback */
+    if (core->isPlaying) {
+        status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+        if (status != kAudioHardwareNoError) {
+            coreaudio_logerr (status, "Can not stop playback\n");
+        }
+        core->isPlaying = 0;
+    }
+
+    /* remove callback */
+    status = AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr (status, "Can not remove IOProc\n");
+    }
+    core->outputDeviceID = kAudioDeviceUnknown;
+
+    /* destroy mutex */
+    err = pthread_mutex_destroy(&core->mutex);
+    if (err) {
+        dolog("Can 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 (!core->isPlaying) {
+            status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+            if (status != kAudioHardwareNoError) {
+                coreaudio_logerr (status, "Can not unpause playback\n");
+            }
+            core->isPlaying = 1;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        /* stop playback */
+        if (core->isPlaying) {
+            status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+            if (status != kAudioHardwareNoError) {
+                coreaudio_logerr (status, "Can not pause playback\n");
+            }
+            core->isPlaying = 0;
+        }
+        break;
+    }
+    return 0;
+}
+
+static void *coreaudio_audio_init (void)
+{
+    return &coreaudio_audio_init;
+}
+
+static void coreaudio_audio_fini (void *opaque)
+{
+    (void) opaque;
+}
+
+static struct audio_option coreaudio_options[] = {
+    {"BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_frames,
+     "Size of the buffer in frames", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops coreaudio_pcm_ops = {
+    coreaudio_init_out,
+    coreaudio_fini_out,
+    coreaudio_run_out,
+    coreaudio_write,
+    coreaudio_ctl_out,
+
+    NULL,
+    NULL,
+    NULL,
+    NULL,
+    NULL
+};
+
+struct audio_driver coreaudio_audio_driver = {
+    INIT_FIELD (name           = ) "coreaudio",
+    INIT_FIELD (descr          = )
+    "CoreAudio http://developer.apple.com/audio/coreaudio.html",
+    INIT_FIELD (options        = ) coreaudio_options,
+    INIT_FIELD (init           = ) coreaudio_audio_init,
+    INIT_FIELD (fini           = ) coreaudio_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &coreaudio_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) 1,
+    INIT_FIELD (max_voices_in  = ) 0,
+    INIT_FIELD (voice_size_out = ) sizeof (coreaudioVoiceOut),
+    INIT_FIELD (voice_size_in  = ) 0
+};
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
new file mode 100644
index 0000000000..a04806eae0
--- /dev/null
+++ b/audio/dsound_template.h
@@ -0,0 +1,298 @@
+/*
+ * 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 TYPE in
+#define IFACE IDirectSoundCaptureBuffer
+#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
+#define FIELD dsound_capture_buffer
+#else
+#define NAME "playback buffer"
+#define TYPE out
+#define IFACE IDirectSoundBuffer
+#define BUFPTR LPDIRECTSOUNDBUFFER
+#define FIELD dsound_buffer
+#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, "Can 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
+    )
+{
+    HRESULT hr;
+    int i;
+    LPVOID p1 = NULL, p2 = NULL;
+    DWORD blen1 = 0, blen2 = 0;
+
+    for (i = 0; i < conf.lock_retries; ++i) {
+        hr = glue (IFACE, _Lock) (
+            buf,
+            pos,
+            len,
+            &p1,
+            &blen1,
+            &p2,
+            &blen2,
+            (entire
+#ifdef DSBTYPE_IN
+             ? DSCBLOCK_ENTIREBUFFER
+#else
+             ? DSBLOCK_ENTIREBUFFER
+#endif
+             : 0)
+            );
+
+        if (FAILED (hr)) {
+#ifndef DSBTYPE_IN
+            if (hr == DSERR_BUFFERLOST) {
+                if (glue (dsound_restore_, TYPE) (buf)) {
+                    dsound_logerr (hr, "Can not lock " NAME "\n");
+                    goto fail;
+                }
+                continue;
+            }
+#endif
+            dsound_logerr (hr, "Can not lock " NAME "\n");
+            goto fail;
+        }
+
+        break;
+    }
+
+    if (i == conf.lock_retries) {
+        dolog ("%d attempts to lock " NAME " failed\n", i);
+        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, "Can not stop " NAME "\n");
+        }
+
+        hr = glue (IFACE, _Release) (ds->FIELD);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not release " NAME "\n");
+        }
+        ds->FIELD = NULL;
+    }
+}
+
+#ifdef DSBTYPE_IN
+static int dsound_init_in (
+    HWVoiceIn *hw,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+#else
+static int dsound_init_out (
+    HWVoiceOut *hw,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+#endif
+{
+    int err;
+    HRESULT hr;
+    dsound *s = &glob_dsound;
+    WAVEFORMATEX wfx;
+    struct full_fmt full_fmt;
+#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
+
+    full_fmt.freq = freq;
+    full_fmt.nchannels = nchannels;
+    full_fmt.fmt = fmt;
+    err = waveformat_from_full_fmt (&wfx, &full_fmt);
+    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, "Can not create " NAME "\n");
+        return -1;
+    }
+
+    hr = glue (IFACE, _GetFormat) (
+        ds->FIELD,
+        &wfx,
+        sizeof (wfx),
+        NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Can 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, "Can not get " NAME " format\n");
+        goto fail0;
+    }
+
+    err = waveformat_to_full_fmt (&wfx, &full_fmt);
+    if (err) {
+        goto fail0;
+    }
+
+    ds->first_time = 1;
+    hw->bufsize = bc.dwBufferBytes;
+    audio_pcm_init_info (
+        &hw->info,
+        full_fmt.freq,
+        full_fmt.nchannels,
+        full_fmt.fmt,
+        audio_need_to_swap_endian (0)
+        );
+
+#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, full_fmt.freq, full_fmt.nchannels, full_fmt.fmt);
+#endif
+    return 0;
+
+ fail0:
+    glue (dsound_fini_, TYPE) (hw);
+    return -1;
+}
+
+#undef NAME
+#undef TYPE
+#undef IFACE
+#undef BUFPTR
+#undef FIELD
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
new file mode 100644
index 0000000000..64b84174d1
--- /dev/null
+++ b/audio/dsoundaudio.c
@@ -0,0 +1,1082 @@
+/*
+ * 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 "vl.h"
+
+#define AUDIO_CAP "dsound"
+#include "audio_int.h"
+
+#include <windows.h>
+#include <objbase.h>
+#include <dsound.h>
+
+/* #define DEBUG_DSOUND */
+
+struct full_fmt {
+    int freq;
+    int nchannels;
+    audfmt_e fmt;
+};
+
+static struct {
+    int lock_retries;
+    int restore_retries;
+    int getstatus_retries;
+    int set_primary;
+    int bufsize_in;
+    int bufsize_out;
+    struct full_fmt full_fmt;
+    int latency_millis;
+} conf = {
+    1,
+    1,
+    1,
+    0,
+    16384,
+    16384,
+    {
+        44100,
+        2,
+        AUD_FMT_S16
+    },
+    10
+};
+
+typedef struct {
+    LPDIRECTSOUND dsound;
+    LPDIRECTSOUNDCAPTURE dsound_capture;
+    LPDIRECTSOUNDBUFFER dsound_primary_buffer;
+    struct full_fmt fmt;
+} dsound;
+
+static dsound glob_dsound;
+
+typedef struct {
+    HWVoiceOut hw;
+    LPDIRECTSOUNDBUFFER dsound_buffer;
+    DWORD old_pos;
+    int first_time;
+#ifdef DEBUG_DSOUND
+    DWORD old_ppos;
+    DWORD played;
+    DWORD mixed;
+#endif
+} DSoundVoiceOut;
+
+typedef struct {
+    HWVoiceIn hw;
+    int first_time;
+    LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
+} 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, "Can 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)
+{
+    HRESULT hr;
+    int i;
+
+    for (i = 0; i < conf.restore_retries; ++i) {
+        hr = IDirectSoundBuffer_Restore (dsb);
+
+        switch (hr) {
+        case DS_OK:
+            return 0;
+
+        case DSERR_BUFFERLOST:
+            continue;
+
+        default:
+            dsound_logerr (hr, "Can not restore playback buffer\n");
+            return -1;
+        }
+    }
+
+    dolog ("%d attempts to restore playback buffer failed\n", i);
+    return -1;
+}
+
+static int waveformat_from_full_fmt (WAVEFORMATEX *wfx,
+                                     struct full_fmt *full_fmt)
+{
+    memset (wfx, 0, sizeof (*wfx));
+
+    wfx->wFormatTag = WAVE_FORMAT_PCM;
+    wfx->nChannels = full_fmt->nchannels;
+    wfx->nSamplesPerSec = full_fmt->freq;
+    wfx->nAvgBytesPerSec = full_fmt->freq << (full_fmt->nchannels == 2);
+    wfx->nBlockAlign = 1 << (full_fmt->nchannels == 2);
+    wfx->cbSize = 0;
+
+    switch (full_fmt->fmt) {
+    case AUD_FMT_S8:
+        wfx->wBitsPerSample = 8;
+        break;
+
+    case AUD_FMT_U8:
+        wfx->wBitsPerSample = 8;
+        break;
+
+    case AUD_FMT_S16:
+        wfx->wBitsPerSample = 16;
+        wfx->nAvgBytesPerSec <<= 1;
+        wfx->nBlockAlign <<= 1;
+        break;
+
+    case AUD_FMT_U16:
+        wfx->wBitsPerSample = 16;
+        wfx->nAvgBytesPerSec <<= 1;
+        wfx->nBlockAlign <<= 1;
+        break;
+
+    default:
+        dolog ("Internal logic error: Bad audio format %d\n",
+               full_fmt->freq);
+        return -1;
+    }
+
+    return 0;
+}
+
+static int waveformat_to_full_fmt (WAVEFORMATEX *wfx,
+                                   struct full_fmt *full_fmt)
+{
+    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;
+    }
+    full_fmt->freq = wfx->nSamplesPerSec;
+
+    switch (wfx->nChannels) {
+    case 1:
+        full_fmt->nchannels = 1;
+        break;
+
+    case 2:
+        full_fmt->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:
+        full_fmt->fmt = AUD_FMT_U8;
+        break;
+
+    case 16:
+        full_fmt->fmt = AUD_FMT_S16;
+        break;
+
+    default:
+        dolog ("Invalid wave format, bits per sample is not 8 or 16, but %d\n",
+               wfx->wBitsPerSample);
+        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)
+{
+    HRESULT hr;
+    int i;
+
+    for (i = 0; i < conf.getstatus_retries; ++i) {
+        hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not get playback buffer status\n");
+            return -1;
+        }
+
+        if (*statusp & DSERR_BUFFERLOST) {
+            if (dsound_restore_out (dsb)) {
+                return -1;
+            }
+            continue;
+        }
+        break;
+    }
+
+    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, "Can 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;
+    st_sample_t *src1 = hw->mix_buf + hw->rpos;
+    st_sample_t *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);
+        mixeng_clear (src1, src_len1);
+    }
+
+    if (src_len2) {
+        dst = advance (dst, src_len1 << hw->info.shift);
+        hw->clip (dst, src2, src_len2);
+        mixeng_clear (src2, src_len2);
+    }
+
+    hw->rpos = pos % hw->samples;
+}
+
+static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)
+{
+    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
+        );
+    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 void dsound_close (dsound *s)
+{
+    HRESULT hr;
+
+    if (s->dsound_primary_buffer) {
+        hr = IDirectSoundBuffer_Release (s->dsound_primary_buffer);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not release primary buffer\n");
+        }
+        s->dsound_primary_buffer = NULL;
+    }
+}
+
+static int dsound_open (dsound *s)
+{
+    int err;
+    HRESULT hr;
+    WAVEFORMATEX wfx;
+    DSBUFFERDESC dsbd;
+    HWND hwnd;
+
+    hwnd = GetForegroundWindow ();
+    hr = IDirectSound_SetCooperativeLevel (
+        s->dsound,
+        hwnd,
+        DSSCL_PRIORITY
+        );
+
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not set cooperative level for window %p\n",
+                       hwnd);
+        return -1;
+    }
+
+    if (!conf.set_primary) {
+        return 0;
+    }
+
+    err = waveformat_from_full_fmt (&wfx, &conf.full_fmt);
+    if (err) {
+        return -1;
+    }
+
+    memset (&dsbd, 0, sizeof (dsbd));
+    dsbd.dwSize = sizeof (dsbd);
+    dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER;
+    dsbd.dwBufferBytes = 0;
+    dsbd.lpwfxFormat = NULL;
+
+    hr = IDirectSound_CreateSoundBuffer (
+        s->dsound,
+        &dsbd,
+        &s->dsound_primary_buffer,
+        NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not create primary playback buffer\n");
+        return -1;
+    }
+
+    hr = IDirectSoundBuffer_SetFormat (s->dsound_primary_buffer, &wfx);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not set primary playback buffer format\n");
+    }
+
+    hr = IDirectSoundBuffer_GetFormat (
+        s->dsound_primary_buffer,
+        &wfx,
+        sizeof (wfx),
+        NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not get primary playback buffer format\n");
+        goto fail0;
+    }
+
+#ifdef DEBUG_DSOUND
+    dolog ("Primary\n");
+    print_wave_format (&wfx);
+#endif
+
+    err = waveformat_to_full_fmt (&wfx, &s->fmt);
+    if (err) {
+        goto fail0;
+    }
+
+    return 0;
+
+ fail0:
+    dsound_close (s);
+    return -1;
+}
+
+static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    HRESULT hr;
+    DWORD status;
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+    LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+
+    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)) {
+            return -1;
+        }
+
+        if (status & DSBSTATUS_PLAYING) {
+            dolog ("warning: voice is already playing\n");
+            return 0;
+        }
+
+        dsound_clear_sample (hw, dsb);
+
+        hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not start playing buffer\n");
+            return -1;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        if (dsound_get_status_out (dsb, &status)) {
+            return -1;
+        }
+
+        if (status & DSBSTATUS_PLAYING) {
+            hr = IDirectSoundBuffer_Stop (dsb);
+            if (FAILED (hr)) {
+                dsound_logerr (hr, "Can 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 err;
+    HRESULT hr;
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+    LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+    int live, len, hwshift;
+    DWORD blen1, blen2;
+    DWORD len1, len2;
+    DWORD decr;
+    DWORD wpos, ppos, old_pos;
+    LPVOID p1, p2;
+
+    if (!dsb) {
+        dolog ("Attempt to run empty with playback buffer\n");
+        return 0;
+    }
+
+    hwshift = hw->info.shift;
+
+    live = audio_pcm_hw_get_live_out (hw);
+
+    hr = IDirectSoundBuffer_GetCurrentPosition (
+        dsb,
+        &ppos,
+        ds->first_time ? &wpos : NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not get playback buffer position\n");
+        return 0;
+    }
+
+    len = live << hwshift;
+
+    if (ds->first_time) {
+        if (conf.latency_millis) {
+            DWORD cur_blat = audio_ring_dist (wpos, ppos, hw->bufsize);
+
+            ds->first_time = 0;
+            old_pos = wpos;
+            old_pos +=
+                millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
+            old_pos %= hw->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 + hw->bufsize))) {
+            len = hw->bufsize - old_pos + ppos;
+        }
+    }
+
+    if (audio_bug (AUDIO_FUNC, len < 0 || len > hw->bufsize)) {
+        dolog ("len=%d hw->bufsize=%d old_pos=%ld ppos=%ld\n",
+               len, hw->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
+        );
+    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)) % hw->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, "Can 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, "Can 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;
+
+    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, "Can 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
+        );
+    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, &nominal_volume);
+    }
+
+    if (p2 && len2) {
+        hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
+    }
+
+    dsound_unlock_in (dscb, p1, p2, blen1, blen2);
+    hw->wpos = (hw->wpos + decr) % hw->samples;
+    return decr;
+}
+
+static void dsound_audio_fini (void *opaque)
+{
+    HRESULT hr;
+    dsound *s = opaque;
+
+    if (!s->dsound) {
+        return;
+    }
+
+    hr = IDirectSound_Release (s->dsound);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not release DirectSound\n");
+    }
+    s->dsound = NULL;
+
+    if (!s->dsound_capture) {
+        return;
+    }
+
+    hr = IDirectSoundCapture_Release (s->dsound_capture);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not release DirectSoundCapture\n");
+    }
+    s->dsound_capture = NULL;
+}
+
+static void *dsound_audio_init (void)
+{
+    int err;
+    HRESULT hr;
+    dsound *s = &glob_dsound;
+
+    hr = CoInitialize (NULL);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not initialize COM\n");
+        return NULL;
+    }
+
+    hr = CoCreateInstance (
+        &CLSID_DirectSound,
+        NULL,
+        CLSCTX_ALL,
+        &IID_IDirectSound,
+        (void **) &s->dsound
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not create DirectSound instance\n");
+        return NULL;
+    }
+
+    hr = IDirectSound_Initialize (s->dsound, NULL);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not initialize DirectSound\n");
+        return NULL;
+    }
+
+    hr = CoCreateInstance (
+        &CLSID_DirectSoundCapture,
+        NULL,
+        CLSCTX_ALL,
+        &IID_IDirectSoundCapture,
+        (void **) &s->dsound_capture
+        );
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not create DirectSoundCapture instance\n");
+    }
+    else {
+        hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not initialize DirectSoundCapture\n");
+
+            hr = IDirectSoundCapture_Release (s->dsound_capture);
+            if (FAILED (hr)) {
+                dsound_logerr (hr, "Can 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[] = {
+    {"LOCK_RETRIES", AUD_OPT_INT, &conf.lock_retries,
+     "Number of times to attempt locking the buffer", NULL, 0},
+    {"RESTOURE_RETRIES", AUD_OPT_INT, &conf.restore_retries,
+     "Number of times to attempt restoring the buffer", NULL, 0},
+    {"GETSTATUS_RETRIES", AUD_OPT_INT, &conf.getstatus_retries,
+     "Number of times to attempt getting status of the buffer", NULL, 0},
+    {"SET_PRIMARY", AUD_OPT_BOOL, &conf.set_primary,
+     "Set the parameters of primary buffer", NULL, 0},
+    {"LATENCY_MILLIS", AUD_OPT_INT, &conf.latency_millis,
+     "(undocumented)", NULL, 0},
+    {"PRIMARY_FREQ", AUD_OPT_INT, &conf.full_fmt.freq,
+     "Primary buffer frequency", NULL, 0},
+    {"PRIMARY_CHANNELS", AUD_OPT_INT, &conf.full_fmt.nchannels,
+     "Primary buffer number of channels (1 - mono, 2 - stereo)", NULL, 0},
+    {"PRIMARY_FMT", AUD_OPT_FMT, &conf.full_fmt.fmt,
+     "Primary buffer format", NULL, 0},
+    {"BUFSIZE_OUT", AUD_OPT_INT, &conf.bufsize_out,
+     "(undocumented)", NULL, 0},
+    {"BUFSIZE_IN", AUD_OPT_INT, &conf.bufsize_in,
+     "(undocumented)", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops dsound_pcm_ops = {
+    dsound_init_out,
+    dsound_fini_out,
+    dsound_run_out,
+    dsound_write,
+    dsound_ctl_out,
+
+    dsound_init_in,
+    dsound_fini_in,
+    dsound_run_in,
+    dsound_read,
+    dsound_ctl_in
+};
+
+struct audio_driver dsound_audio_driver = {
+    INIT_FIELD (name           = ) "dsound",
+    INIT_FIELD (descr          = )
+    "DirectSound http://wikipedia.org/wiki/DirectSound",
+    INIT_FIELD (options        = ) dsound_options,
+    INIT_FIELD (init           = ) dsound_audio_init,
+    INIT_FIELD (fini           = ) dsound_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &dsound_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) 1,
+    INIT_FIELD (voice_size_out = ) sizeof (DSoundVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (DSoundVoiceIn)
+};
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
index 7b79026a82..36b8d47c43 100644
--- a/audio/fmodaudio.c
+++ b/audio/fmodaudio.c
@@ -1,8 +1,8 @@
 /*
- * QEMU FMOD audio output driver
- * 
- * Copyright (c) 2004 Vassili Karpov (malc)
- * 
+ * QEMU FMOD 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
@@ -25,53 +25,77 @@
 #include <fmod_errors.h>
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "fmod"
+#include "audio_int.h"
 
-typedef struct FMODVoice {
-    HWVoice hw;
+typedef struct FMODVoiceOut {
+    HWVoiceOut hw;
     unsigned int old_pos;
     FSOUND_SAMPLE *fmod_sample;
     int channel;
-} FMODVoice;
-
-#define dolog(...) AUD_log ("fmod", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+} FMODVoiceOut;
 
-#define QC_FMOD_DRV "QEMU_FMOD_DRV"
-#define QC_FMOD_FREQ "QEMU_FMOD_FREQ"
-#define QC_FMOD_SAMPLES "QEMU_FMOD_SAMPLES"
-#define QC_FMOD_CHANNELS "QEMU_FMOD_CHANNELS"
-#define QC_FMOD_BUFSIZE "QEMU_FMOD_BUFSIZE"
-#define QC_FMOD_THRESHOLD "QEMU_FMOD_THRESHOLD"
+typedef struct FMODVoiceIn {
+    HWVoiceIn hw;
+    FSOUND_SAMPLE *fmod_sample;
+} FMODVoiceIn;
 
 static struct {
+    const char *drvname;
     int nb_samples;
     int freq;
     int nb_channels;
     int bufsize;
     int threshold;
+    int broken_adc;
 } conf = {
-    2048,
+    NULL,
+    2048 * 2,
     44100,
-    1,
+    2,
+    0,
     0,
-    128
+    0
 };
 
-#define errstr() FMOD_ErrorString (FSOUND_GetError ())
+static void GCC_FMT_ATTR (1, 2) fmod_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",
+             FMOD_ErrorString (FSOUND_GetError ()));
+}
+
+static void GCC_FMT_ATTR (2, 3) fmod_logerr2 (
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Can 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",
+             FMOD_ErrorString (FSOUND_GetError ()));
+}
 
-static int fmod_hw_write (SWVoice *sw, void *buf, int len)
+static int fmod_write (SWVoiceOut *sw, void *buf, int len)
 {
-    return pcm_hw_write (sw, buf, len);
+    return audio_pcm_sw_write (sw, buf, len);
 }
 
-static void fmod_clear_sample (FMODVoice *fmd)
+static void fmod_clear_sample (FMODVoiceOut *fmd)
 {
-    HWVoice *hw = &fmd->hw;
+    HWVoiceOut *hw = &fmd->hw;
     int status;
     void *p1 = 0, *p2 = 0;
     unsigned int len1 = 0, len2 = 0;
@@ -79,7 +103,7 @@ static void fmod_clear_sample (FMODVoice *fmd)
     status = FSOUND_Sample_Lock (
         fmd->fmod_sample,
         0,
-        hw->samples << hw->shift,
+        hw->samples << hw->info.shift,
         &p1,
         &p2,
         &len1,
@@ -87,78 +111,88 @@ static void fmod_clear_sample (FMODVoice *fmd)
         );
 
     if (!status) {
-        dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to lock sample\n");
         return;
     }
 
-    if ((len1 & hw->align) || (len2 & hw->align)) {
-        dolog ("Locking sample returned unaligned length %d, %d\n",
-               len1, len2);
+    if ((len1 & hw->info.align) || (len2 & hw->info.align)) {
+        dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
+               len1, len2, hw->info.align + 1);
         goto fail;
     }
 
-    if (len1 + len2 != hw->samples << hw->shift) {
-        dolog ("Locking sample returned incomplete length %d, %d\n",
-               len1 + len2, hw->samples << hw->shift);
+    if ((len1 + len2) - (hw->samples << hw->info.shift)) {
+        dolog ("Lock returned incomplete length %d, %d\n",
+               len1 + len2, hw->samples << hw->info.shift);
         goto fail;
     }
-    pcm_hw_clear (hw, p1, hw->samples);
+
+    audio_pcm_info_clear_buf (&hw->info, p1, hw->samples);
 
  fail:
     status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, len1, len2);
     if (!status) {
-        dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to unlock sample\n");
     }
 }
 
-static int fmod_write_sample (HWVoice *hw, uint8_t *dst, st_sample_t *src,
-                              int src_size, int src_pos, int dst_len)
+static void fmod_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
 {
-    int src_len1 = dst_len, src_len2 = 0, pos = src_pos + dst_len;
-    st_sample_t *src1 = src + src_pos, *src2 = 0;
-
-    if (src_pos + dst_len > src_size) {
-        src_len1 = src_size - src_pos;
-        src2 = src;
+    int src_len1 = dst_len;
+    int src_len2 = 0;
+    int pos = hw->rpos + dst_len;
+    st_sample_t *src1 = hw->mix_buf + hw->rpos;
+    st_sample_t *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);
-        memset (src1, 0, src_len1 * sizeof (st_sample_t));
-        advance (dst, src_len1);
+        mixeng_clear (src1, src_len1);
     }
 
     if (src_len2) {
+        dst = advance (dst, src_len1 << hw->info.shift);
         hw->clip (dst, src2, src_len2);
-        memset (src2, 0, src_len2 * sizeof (st_sample_t));
+        mixeng_clear (src2, src_len2);
     }
-    return pos;
+
+    hw->rpos = pos % hw->samples;
 }
 
-static int fmod_unlock_sample (FMODVoice *fmd, void *p1, void *p2,
+static int fmod_unlock_sample (FSOUND_SAMPLE *sample, void *p1, void *p2,
                                unsigned int blen1, unsigned int blen2)
 {
-    int status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, blen1, blen2);
+    int status = FSOUND_Sample_Unlock (sample, p1, p2, blen1, blen2);
     if (!status) {
-        dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to unlock sample\n");
         return -1;
     }
     return 0;
 }
 
-static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
-                             void **p1, void **p2,
-                             unsigned int *blen1, unsigned int *blen2)
+static int fmod_lock_sample (
+    FSOUND_SAMPLE *sample,
+    struct audio_pcm_info *info,
+    int pos,
+    int len,
+    void **p1,
+    void **p2,
+    unsigned int *blen1,
+    unsigned int *blen2
+    )
 {
-    HWVoice *hw = &fmd->hw;
     int status;
 
     status = FSOUND_Sample_Lock (
-        fmd->fmod_sample,
-        pos << hw->shift,
-        len << hw->shift,
+        sample,
+        pos << info->shift,
+        len << info->shift,
         p1,
         p2,
         blen1,
@@ -166,89 +200,117 @@ static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
         );
 
     if (!status) {
-        dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to lock sample\n");
         return -1;
     }
 
-    if ((*blen1 & hw->align) || (*blen2 & hw->align)) {
-        dolog ("Locking sample returned unaligned length %d, %d\n",
-               *blen1, *blen2);
-        fmod_unlock_sample (fmd, *p1, *p2, *blen1, *blen2);
+    if ((*blen1 & info->align) || (*blen2 & info->align)) {
+        dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
+               *blen1, *blen2, info->align + 1);
+
+        fmod_unlock_sample (sample, *p1, *p2, *blen1, *blen2);
+
+        *p1 = NULL - 1;
+        *p2 = NULL - 1;
+        *blen1 = ~0U;
+        *blen2 = ~0U;
         return -1;
     }
+
+    if (!*p1 && *blen1) {
+        dolog ("warning: !p1 && blen1=%d\n", *blen1);
+        *blen1 = 0;
+    }
+
+    if (!p2 && *blen2) {
+        dolog ("warning: !p2 && blen2=%d\n", *blen2);
+        *blen2 = 0;
+    }
+
     return 0;
 }
 
-static void fmod_hw_run (HWVoice *hw)
+static int fmod_run_out (HWVoiceOut *hw)
 {
-    FMODVoice *fmd = (FMODVoice *) hw;
-    int rpos, live, decr;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
+    int live, decr;
     void *p1 = 0, *p2 = 0;
     unsigned int blen1 = 0, blen2 = 0;
     unsigned int len1 = 0, len2 = 0;
-    int nb_active;
+    int nb_live;
 
-    live = pcm_hw_get_live2 (hw, &nb_active);
-    if (live <= 0) {
-        return;
+    live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
+    if (!live) {
+        return 0;
     }
 
     if (!hw->pending_disable
-        && nb_active
-        && conf.threshold
-        && live <= conf.threshold) {
-        ldebug ("live=%d nb_active=%d\n", live, nb_active);
-        return;
+        && nb_live
+        && (conf.threshold && live <= conf.threshold)) {
+        ldebug ("live=%d nb_live=%d\n", live, nb_live);
+        return 0;
     }
 
     decr = live;
 
-#if 1
     if (fmd->channel >= 0) {
-        int pos2 = (fmd->old_pos + decr) % hw->samples;
-        int pos = FSOUND_GetCurrentPosition (fmd->channel);
+        int len = decr;
+        int old_pos = fmd->old_pos;
+        int ppos = FSOUND_GetCurrentPosition (fmd->channel);
 
-        if (fmd->old_pos < pos && pos2 >= pos) {
-            decr = pos - fmd->old_pos - (pos2 == pos) - 1;
+        if (ppos == old_pos || !ppos) {
+            return 0;
         }
-        else if (fmd->old_pos > pos && pos2 >= pos && pos2 < fmd->old_pos) {
-            decr = (hw->samples - fmd->old_pos) + pos - (pos2 == pos) - 1;
+
+        if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
+            len = ppos - old_pos;
+        }
+        else {
+            if ((old_pos > ppos) && ((old_pos + len) > (ppos + hw->samples))) {
+                len = hw->samples - old_pos + ppos;
+            }
+        }
+        decr = len;
+
+        if (audio_bug (AUDIO_FUNC, decr < 0)) {
+            dolog ("decr=%d live=%d ppos=%d old_pos=%d len=%d\n",
+                   decr, live, ppos, old_pos, len);
+            return 0;
         }
-/*         ldebug ("pos=%d pos2=%d old=%d live=%d decr=%d\n", */
-/*                 pos, pos2, fmd->old_pos, live, decr); */
     }
-#endif
 
-    if (decr <= 0) {
-        return;
+
+    if (!decr) {
+        return 0;
     }
 
-    if (fmod_lock_sample (fmd, fmd->old_pos, decr, &p1, &p2, &blen1, &blen2)) {
-        return;
+    if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
+                          fmd->old_pos, decr,
+                          &p1, &p2,
+                          &blen1, &blen2)) {
+        return 0;
     }
 
-    len1 = blen1 >> hw->shift;
-    len2 = blen2 >> hw->shift;
+    len1 = blen1 >> hw->info.shift;
+    len2 = blen2 >> hw->info.shift;
     ldebug ("%p %p %d %d %d %d\n", p1, p2, len1, len2, blen1, blen2);
     decr = len1 + len2;
-    rpos = hw->rpos;
 
-    if (len1) {
-        rpos = fmod_write_sample (hw, p1, hw->mix_buf, hw->samples, rpos, len1);
+    if (p1 && len1) {
+        fmod_write_sample (hw, p1, len1);
     }
 
-    if (len2) {
-        rpos = fmod_write_sample (hw, p2, hw->mix_buf, hw->samples, rpos, len2);
+    if (p2 && len2) {
+        fmod_write_sample (hw, p2, len2);
     }
 
-    fmod_unlock_sample (fmd, p1, p2, blen1, blen2);
+    fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
 
-    pcm_hw_dec_live (hw, decr);
-    hw->rpos = rpos % hw->samples;
     fmd->old_pos = (fmd->old_pos + decr) % hw->samples;
+    return decr;
 }
 
-static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
+static int aud_to_fmodfmt (audfmt_e fmt, int stereo)
 {
     int mode = FSOUND_LOOP_NORMAL;
 
@@ -270,16 +332,19 @@ static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
         break;
 
     default:
-        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
-        exit (EXIT_FAILURE);
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_FMOD
+        abort ();
+#endif
+        mode |= FSOUND_8BITS;
     }
     mode |= stereo ? FSOUND_STEREO : FSOUND_MONO;
     return mode;
 }
 
-static void fmod_hw_fini (HWVoice *hw)
+static void fmod_fini_out (HWVoiceOut *hw)
 {
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
 
     if (fmd->fmod_sample) {
         FSOUND_Sample_Free (fmd->fmod_sample);
@@ -291,12 +356,12 @@ static void fmod_hw_fini (HWVoice *hw)
     }
 }
 
-static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int fmod_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
     int bits16, mode, channel;
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
 
-    mode = AUD_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+    mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
     fmd->fmod_sample = FSOUND_Sample_Alloc (
         FSOUND_FREE,            /* index */
         conf.nb_samples,        /* length */
@@ -308,52 +373,145 @@ static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
         );
 
     if (!fmd->fmod_sample) {
-        dolog ("Failed to allocate FMOD sample\nReason: %s\n", errstr ());
+        fmod_logerr2 ("DAC", "Failed to allocate FMOD sample\n");
         return -1;
     }
 
     channel = FSOUND_PlaySoundEx (FSOUND_FREE, fmd->fmod_sample, 0, 1);
     if (channel < 0) {
-        dolog ("Failed to start playing sound\nReason: %s\n", errstr ());
+        fmod_logerr2 ("DAC", "Failed to start playing sound\n");
         FSOUND_Sample_Free (fmd->fmod_sample);
         return -1;
     }
     fmd->channel = channel;
 
-    hw->freq = freq;
-    hw->fmt = fmt;
-    hw->nchannels = nchannels;
-    bits16 = fmt == AUD_FMT_U16 || fmt == AUD_FMT_S16;
+    /* FMOD always operates on little endian frames? */
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
+                         audio_need_to_swap_endian (0));
+    bits16 = (mode & FSOUND_16BITS) != 0;
     hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
     return 0;
 }
 
-static int fmod_hw_ctl (HWVoice *hw, int cmd, ...)
+static int fmod_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     int status;
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
 
     switch (cmd) {
     case VOICE_ENABLE:
         fmod_clear_sample (fmd);
         status = FSOUND_SetPaused (fmd->channel, 0);
         if (!status) {
-            dolog ("Failed to resume channel %d\nReason: %s\n",
-                   fmd->channel, errstr ());
+            fmod_logerr ("Failed to resume channel %d\n", fmd->channel);
         }
         break;
 
     case VOICE_DISABLE:
         status = FSOUND_SetPaused (fmd->channel, 1);
         if (!status) {
-            dolog ("Failed to pause channel %d\nReason: %s\n",
-                   fmd->channel, errstr ());
+            fmod_logerr ("Failed to pause channel %d\n", fmd->channel);
         }
         break;
     }
     return 0;
 }
 
+static int fmod_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    int bits16, mode;
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+    if (conf.broken_adc) {
+        return -1;
+    }
+
+    mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+    fmd->fmod_sample = FSOUND_Sample_Alloc (
+        FSOUND_FREE,            /* index */
+        conf.nb_samples,        /* length */
+        mode,                   /* mode */
+        freq,                   /* freq */
+        255,                    /* volume */
+        128,                    /* pan */
+        255                     /* priority */
+        );
+
+    if (!fmd->fmod_sample) {
+        fmod_logerr2 ("ADC", "Failed to allocate FMOD sample\n");
+        return -1;
+    }
+
+    /* FMOD always operates on little endian frames? */
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
+                         audio_need_to_swap_endian (0));
+    bits16 = (mode & FSOUND_16BITS) != 0;
+    hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
+    return 0;
+}
+
+static void fmod_fini_in (HWVoiceIn *hw)
+{
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+    if (fmd->fmod_sample) {
+        FSOUND_Record_Stop ();
+        FSOUND_Sample_Free (fmd->fmod_sample);
+        fmd->fmod_sample = 0;
+    }
+}
+
+static int fmod_run_in (HWVoiceIn *hw)
+{
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+    int hwshift = hw->info.shift;
+    int live, dead, new_pos, len;
+    unsigned int blen1 = 0, blen2 = 0;
+    unsigned int len1, len2;
+    unsigned int decr;
+    void *p1, *p2;
+
+    live = audio_pcm_hw_get_live_in (hw);
+    dead = hw->samples - live;
+    if (!dead) {
+        return 0;
+    }
+
+    new_pos = FSOUND_Record_GetPosition ();
+    if (new_pos < 0) {
+        fmod_logerr ("Can not get recording position\n");
+        return 0;
+    }
+
+    len = audio_ring_dist (new_pos,  hw->wpos, hw->samples);
+    if (!len) {
+        return 0;
+    }
+    len = audio_MIN (len, dead);
+
+    if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
+                          hw->wpos, len,
+                          &p1, &p2,
+                          &blen1, &blen2)) {
+        return 0;
+    }
+
+    len1 = blen1 >> hwshift;
+    len2 = blen2 >> hwshift;
+    decr = len1 + len2;
+
+    if (p1 && blen1) {
+        hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
+    }
+    if (p2 && len2) {
+        hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
+    }
+
+    fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
+    hw->wpos = (hw->wpos + decr) % hw->samples;
+    return decr;
+}
+
 static struct {
     const char *name;
     int type;
@@ -378,16 +536,16 @@ static struct {
     {"ps2", FSOUND_OUTPUT_PS2},
     {"gcube", FSOUND_OUTPUT_GC},
 #endif
-    {"nort", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
+    {"none-realtime", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
 };
 
 static void *fmod_audio_init (void)
 {
-    int i;
+    size_t i;
     double ver;
     int status;
     int output_type = -1;
-    const char *drv = audio_get_conf_str (QC_FMOD_DRV, NULL);
+    const char *drv = conf.drvname;
 
     ver = FSOUND_GetVersion ();
     if (ver < FMOD_VERSION) {
@@ -395,6 +553,14 @@ static void *fmod_audio_init (void)
         return NULL;
     }
 
+#ifdef __linux__
+    if (ver < 3.75) {
+        dolog ("FMOD before 3.75 has bug preventing ADC from working\n"
+               "ADC will be disabled.\n");
+        conf.broken_adc = 1;
+    }
+#endif
+
     if (drv) {
         int found = 0;
         for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
@@ -405,65 +571,115 @@ static void *fmod_audio_init (void)
             }
         }
         if (!found) {
-            dolog ("Unknown FMOD output driver `%s'\n", drv);
+            dolog ("Unknown FMOD driver `%s'\n", drv);
+            dolog ("Valid drivers:\n");
+            for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+                dolog ("  %s\n", drvtab[i].name);
+            }
         }
     }
 
     if (output_type != -1) {
         status = FSOUND_SetOutput (output_type);
         if (!status) {
-            dolog ("FSOUND_SetOutput(%d) failed\nReason: %s\n",
-                   output_type, errstr ());
+            fmod_logerr ("FSOUND_SetOutput(%d) failed\n", output_type);
             return NULL;
         }
     }
 
-    conf.freq = audio_get_conf_int (QC_FMOD_FREQ, conf.freq);
-    conf.nb_samples = audio_get_conf_int (QC_FMOD_SAMPLES, conf.nb_samples);
-    conf.nb_channels =
-        audio_get_conf_int (QC_FMOD_CHANNELS,
-                            (audio_state.nb_hw_voices > 1
-                             ? audio_state.nb_hw_voices
-                             : conf.nb_channels));
-    conf.bufsize = audio_get_conf_int (QC_FMOD_BUFSIZE, conf.bufsize);
-    conf.threshold = audio_get_conf_int (QC_FMOD_THRESHOLD, conf.threshold);
-
     if (conf.bufsize) {
         status = FSOUND_SetBufferSize (conf.bufsize);
         if (!status) {
-            dolog ("FSOUND_SetBufferSize (%d) failed\nReason: %s\n",
-                   conf.bufsize, errstr ());
+            fmod_logerr ("FSOUND_SetBufferSize (%d) failed\n", conf.bufsize);
         }
     }
 
     status = FSOUND_Init (conf.freq, conf.nb_channels, 0);
     if (!status) {
-        dolog ("FSOUND_Init failed\nReason: %s\n", errstr ());
+        fmod_logerr ("FSOUND_Init failed\n");
         return NULL;
     }
 
     return &conf;
 }
 
+static int fmod_read (SWVoiceIn *sw, void *buf, int size)
+{
+    return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int fmod_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    int status;
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        status = FSOUND_Record_StartSample (fmd->fmod_sample, 1);
+        if (!status) {
+            fmod_logerr ("Failed to start recording\n");
+        }
+        break;
+
+    case VOICE_DISABLE:
+        status = FSOUND_Record_Stop ();
+        if (!status) {
+            fmod_logerr ("Failed to stop recording\n");
+        }
+        break;
+    }
+    return 0;
+}
+
 static void fmod_audio_fini (void *opaque)
 {
+    (void) opaque;
     FSOUND_Close ();
 }
 
-struct pcm_ops fmod_pcm_ops = {
-    fmod_hw_init,
-    fmod_hw_fini,
-    fmod_hw_run,
-    fmod_hw_write,
-    fmod_hw_ctl
+static struct audio_option fmod_options[] = {
+    {"DRV", AUD_OPT_STR, &conf.drvname,
+     "FMOD driver", NULL, 0},
+    {"FREQ", AUD_OPT_INT, &conf.freq,
+     "Default frequency", NULL, 0},
+    {"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
+     "Buffer size in samples", NULL, 0},
+    {"CHANNELS", AUD_OPT_INT, &conf.nb_channels,
+     "Number of default channels (1 - mono, 2 - stereo)", NULL, 0},
+    {"BUFSIZE", AUD_OPT_INT, &conf.bufsize,
+     "(undocumented)", NULL, 0},
+#if 0
+    {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+     "(undocumented)"},
+#endif
+
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops fmod_pcm_ops = {
+    fmod_init_out,
+    fmod_fini_out,
+    fmod_run_out,
+    fmod_write,
+    fmod_ctl_out,
+
+    fmod_init_in,
+    fmod_fini_in,
+    fmod_run_in,
+    fmod_read,
+    fmod_ctl_in
 };
 
-struct audio_output_driver fmod_output_driver = {
-    "fmod",
-    fmod_audio_init,
-    fmod_audio_fini,
-    &fmod_pcm_ops,
-    1,
-    INT_MAX,
-    sizeof (FMODVoice)
+struct audio_driver fmod_audio_driver = {
+    INIT_FIELD (name           = ) "fmod",
+    INIT_FIELD (descr          = ) "FMOD 3.xx http://www.fmod.org",
+    INIT_FIELD (options        = ) fmod_options,
+    INIT_FIELD (init           = ) fmod_audio_init,
+    INIT_FIELD (fini           = ) fmod_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &fmod_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) INT_MAX,
+    INIT_FIELD (voice_size_out = ) sizeof (FMODVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (FMODVoiceIn)
 };
diff --git a/audio/mixeng.c b/audio/mixeng.c
index b0bb412c63..d43c5e59d7 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -1,7 +1,7 @@
 /*
  * QEMU Mixing engine
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * 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
@@ -23,87 +23,174 @@
  * THE SOFTWARE.
  */
 #include "vl.h"
-//#define DEBUG_FP
-#include "audio/mixeng.h"
 
+#define AUDIO_CAP "mixeng"
+#include "audio_int.h"
+
+#define NOVOL
+
+/* 8 bit */
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+
+/* Signed 8 bit */
 #define IN_T int8_t
-#define IN_MIN CHAR_MIN
-#define IN_MAX CHAR_MAX
+#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 IN_T
+#undef SHIFT
 
+/* Unsigned 8 bit */
 #define IN_T uint8_t
 #define IN_MIN 0
 #define IN_MAX UCHAR_MAX
+#define SHIFT 8
 #include "mixeng_template.h"
 #undef IN_MAX
 #undef IN_MIN
 #undef IN_T
+#undef SHIFT
+
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
 
+/* Signed 16 bit */
 #define IN_T int16_t
 #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 IN_T
+#undef SHIFT
 
 #define IN_T uint16_t
 #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 IN_T
+#undef SHIFT
 
-t_sample *mixeng_conv[2][2][2] = {
+t_sample *mixeng_conv[2][2][2][2] = {
     {
         {
-            conv_uint8_t_to_mono,
-            conv_uint16_t_to_mono
+            {
+                conv_natural_uint8_t_to_mono,
+                conv_natural_uint16_t_to_mono
+            },
+            {
+                conv_natural_uint8_t_to_mono,
+                conv_swap_uint16_t_to_mono
+            }
         },
         {
-            conv_int8_t_to_mono,
-            conv_int16_t_to_mono
+            {
+                conv_natural_int8_t_to_mono,
+                conv_natural_int16_t_to_mono
+            },
+            {
+                conv_natural_int8_t_to_mono,
+                conv_swap_int16_t_to_mono
+            }
         }
     },
     {
         {
-            conv_uint8_t_to_stereo,
-            conv_uint16_t_to_stereo
+            {
+                conv_natural_uint8_t_to_stereo,
+                conv_natural_uint16_t_to_stereo
+            },
+            {
+                conv_natural_uint8_t_to_stereo,
+                conv_swap_uint16_t_to_stereo
+            }
         },
         {
-            conv_int8_t_to_stereo,
-            conv_int16_t_to_stereo
+            {
+                conv_natural_int8_t_to_stereo,
+                conv_natural_int16_t_to_stereo
+            },
+            {
+                conv_natural_int8_t_to_stereo,
+                conv_swap_int16_t_to_stereo
+            }
         }
     }
 };
 
-f_sample *mixeng_clip[2][2][2] = {
+f_sample *mixeng_clip[2][2][2][2] = {
     {
         {
-            clip_uint8_t_from_mono,
-            clip_uint16_t_from_mono
+            {
+                clip_natural_uint8_t_from_mono,
+                clip_natural_uint16_t_from_mono
+            },
+            {
+                clip_natural_uint8_t_from_mono,
+                clip_swap_uint16_t_from_mono
+            }
         },
         {
-            clip_int8_t_from_mono,
-            clip_int16_t_from_mono
+            {
+                clip_natural_int8_t_from_mono,
+                clip_natural_int16_t_from_mono
+            },
+            {
+                clip_natural_int8_t_from_mono,
+                clip_swap_int16_t_from_mono
+            }
         }
     },
     {
         {
-            clip_uint8_t_from_stereo,
-            clip_uint16_t_from_stereo
+            {
+                clip_natural_uint8_t_from_stereo,
+                clip_natural_uint16_t_from_stereo
+            },
+            {
+                clip_natural_uint8_t_from_stereo,
+                clip_swap_uint16_t_from_stereo
+            }
         },
         {
-            clip_int8_t_from_stereo,
-            clip_int16_t_from_stereo
+            {
+                clip_natural_int8_t_from_stereo,
+                clip_natural_int16_t_from_stereo
+            },
+            {
+                clip_natural_int8_t_from_stereo,
+                clip_swap_int16_t_from_stereo
+            }
         }
     }
 };
@@ -116,9 +203,9 @@ f_sample *mixeng_clip[2][2][2] = {
  * 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.  
+ * any purpose.  This copyright notice must be maintained.
+ * Lance Norskog And Sundry Contributors are not responsible for
+ * the consequences of using this software.
  */
 
 /*
@@ -156,21 +243,13 @@ void *st_rate_start (int inrate, int outrate)
     rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
 
     if (!rate) {
-        exit (EXIT_FAILURE);
-    }
-
-    if (inrate == outrate) {
-        // exit (EXIT_FAILURE);
-    }
-
-    if (inrate >= 65535 || outrate >= 65535) {
-        // exit (EXIT_FAILURE);
+        return NULL;
     }
 
     rate->opos = 0;
 
     /* increment */
-    rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate;
+    rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
 
     rate->ipos = 0;
     rate->ilast.l = 0;
@@ -178,78 +257,20 @@ void *st_rate_start (int inrate, int outrate)
     return rate;
 }
 
-/*
- * Processed signed long samples from ibuf to obuf.
- * Return number of samples processed.
- */
-void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
-                   int *isamp, int *osamp)
-{
-    rate_t rate = (rate_t) opaque;
-    st_sample_t *istart, *iend;
-    st_sample_t *ostart, *oend;
-    st_sample_t ilast, icur, out;
-    int64_t t;
-
-    ilast = rate->ilast;
-
-    istart = ibuf;
-    iend = ibuf + *isamp;
+#define NAME st_rate_flow_mix
+#define OP(a, b) a += b
+#include "rate_template.h"
 
-    ostart = obuf;
-    oend = obuf + *osamp;
-
-    if (rate->opos_inc == 1ULL << 32) {
-        int i, n = *isamp > *osamp ? *osamp : *isamp;
-        for (i = 0; i < n; i++) {
-            obuf[i].l += ibuf[i].r;
-            obuf[i].r += ibuf[i].r;
-        }
-        *isamp = n;
-        *osamp = n;
-        return;
-    }
-
-    while (obuf < oend) {
-
-        /* Safety catch to make sure we have input samples.  */
-        if (ibuf >= iend)
-            break;
-
-        /* read as many input samples so that ipos > opos */
-
-        while (rate->ipos <= (rate->opos >> 32)) {
-            ilast = *ibuf++;
-            rate->ipos++;
-            /* See if we finished the input buffer yet */
-            if (ibuf >= iend) goto the_end;
-        }
-
-        icur = *ibuf;
-
-        /* interpolate */
-        t = rate->opos & 0xffffffff;
-        out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX;
-        out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX;
-
-        /* output sample & increment position */
-#if 0
-        *obuf++ = out;
-#else
-        obuf->l += out.l;
-        obuf->r += out.r;
-        obuf += 1;
-#endif
-        rate->opos += rate->opos_inc;
-    }
-
-the_end:
-    *isamp = ibuf - istart;
-    *osamp = obuf - ostart;
-    rate->ilast = ilast;
-}
+#define NAME st_rate_flow
+#define OP(a, b) a = b
+#include "rate_template.h"
 
 void st_rate_stop (void *opaque)
 {
     qemu_free (opaque);
 }
+
+void mixeng_clear (st_sample_t *buf, int len)
+{
+    memset (buf, 0, len * sizeof (st_sample_t));
+}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 699435ea25..9e3bac1744 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -1,8 +1,8 @@
 /*
  * QEMU Mixing engine header
- * 
- * Copyright (c) 2004 Vassili Karpov (malc)
- * 
+ *
+ * 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
@@ -24,16 +24,28 @@
 #ifndef QEMU_MIXENG_H
 #define QEMU_MIXENG_H
 
-typedef void (t_sample) (void *dst, const void *src, int samples);
-typedef void (f_sample) (void *dst, const void *src, int samples);
+#ifdef FLOAT_MIXENG
+typedef float real_t;
+typedef struct { int mute; real_t r; real_t l; } volume_t;
+typedef struct { real_t l; real_t r; } st_sample_t;
+#else
+typedef struct { int mute; int64_t r; int64_t l; } volume_t;
 typedef struct { int64_t l; int64_t r; } st_sample_t;
+#endif
+
+typedef void (t_sample) (st_sample_t *dst, const void *src,
+                         int samples, volume_t *vol);
+typedef void (f_sample) (void *dst, const st_sample_t *src, int samples);
 
-extern t_sample *mixeng_conv[2][2][2];
-extern f_sample *mixeng_clip[2][2][2];
+extern t_sample *mixeng_conv[2][2][2][2];
+extern f_sample *mixeng_clip[2][2][2][2];
 
 void *st_rate_start (int inrate, int outrate);
 void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
                    int *isamp, int *osamp);
+void st_rate_flow_mix (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+                       int *isamp, int *osamp);
 void st_rate_stop (void *opaque);
+void mixeng_clear (st_sample_t *buf, int len);
 
 #endif  /* mixeng.h */
diff --git a/audio/mixeng_template.h b/audio/mixeng_template.h
index f3b3f654fd..d726441e2e 100644
--- a/audio/mixeng_template.h
+++ b/audio/mixeng_template.h
@@ -1,8 +1,8 @@
 /*
  * QEMU Mixing engine
- * 
- * Copyright (c) 2004 Vassili Karpov (malc)
- * 
+ *
+ * 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
@@ -27,85 +27,151 @@
  * dec++'ified by Dscho
  */
 
+#ifndef SIGNED
+#define HALF (IN_MAX >> 1)
+#endif
+
+#ifdef NOVOL
+#define VOL(a, b) a
+#else
+#ifdef FLOAT_MIXENG
+#define VOL(a, b) ((a) * (b))
+#else
+#define VOL(a, b) ((a) * (b)) >> 32
+#endif
+#endif
+
+#define ET glue (ENDIAN_CONVERSION, glue (_, IN_T))
+
+#ifdef FLOAT_MIXENG
+static real_t inline glue (conv_, ET) (IN_T v)
+{
+    IN_T nv = ENDIAN_CONVERT (v);
+
+#ifdef RECIPROCAL
+#ifdef SIGNED
+    return nv * (1.f / (real_t) (IN_MAX - IN_MIN));
+#else
+    return (nv - HALF) * (1.f / (real_t) IN_MAX);
+#endif
+#else  /* !RECIPROCAL */
 #ifdef SIGNED
-#define HALFT IN_MAX
-#define HALF IN_MAX
+    return nv / (real_t) (IN_MAX - IN_MIN);
 #else
-#define HALFT ((IN_MAX)>>1)
-#define HALF HALFT
+    return (nv - HALF) / (real_t) IN_MAX;
 #endif
+#endif
+}
 
-static int64_t inline glue(conv_,IN_T) (IN_T v)
+static IN_T inline glue (clip_, ET) (real_t v)
 {
+    if (v >= 0.5) {
+        return IN_MAX;
+    }
+    else if (v < -0.5) {
+        return IN_MIN;
+    }
+
+#ifdef SIGNED
+    return ENDIAN_CONVERT ((IN_T) (v * (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 (INT_MAX*(int64_t)v)/HALF;
+    return ((int64_t) nv) << (32 - SHIFT);
 #else
-    return (INT_MAX*((int64_t)v-HALFT))/HALF;
+    return ((int64_t) nv - HALF) << (32 - SHIFT);
 #endif
 }
 
-static IN_T inline glue(clip_,IN_T) (int64_t v)
+static inline IN_T glue (clip_, ET) (int64_t v)
 {
-    if (v >= INT_MAX)
+    if (v >= 0x7f000000) {
         return IN_MAX;
-    else if (v < -INT_MAX)
+    }
+    else if (v < -2147483648LL) {
         return IN_MIN;
+    }
 
 #ifdef SIGNED
-    return (IN_T) (v*HALF/INT_MAX);
+    return ENDIAN_CONVERT ((IN_T) (v >> (32 - SHIFT)));
 #else
-    return (IN_T) (v+INT_MAX/2)*HALF/INT_MAX;
+    return ENDIAN_CONVERT ((IN_T) ((v >> (32 - SHIFT)) + HALF));
 #endif
 }
+#endif
 
-static void glue(glue(conv_,IN_T),_to_stereo) (void *dst, const void *src,
-                                               int samples)
+static void glue (glue (conv_, ET), _to_stereo)
+    (st_sample_t *dst, const void *src, int samples, volume_t *vol)
 {
-    st_sample_t *out = (st_sample_t *) dst;
+    st_sample_t *out = dst;
     IN_T *in = (IN_T *) src;
+#ifndef NOVOL
+    if (vol->mute) {
+        mixeng_clear (dst, samples);
+        return;
+    }
+#else
+    (void) vol;
+#endif
     while (samples--) {
-        out->l = glue(conv_,IN_T) (*in++);
-        out->r = glue(conv_,IN_T) (*in++);
+        out->l = VOL (glue (conv_, ET) (*in++), vol->l);
+        out->r = VOL (glue (conv_, ET) (*in++), vol->r);
         out += 1;
     }
 }
 
-static void glue(glue(conv_,IN_T),_to_mono) (void *dst, const void *src,
-                                             int samples)
+static void glue (glue (conv_, ET), _to_mono)
+    (st_sample_t *dst, const void *src, int samples, volume_t *vol)
 {
-    st_sample_t *out = (st_sample_t *) dst;
+    st_sample_t *out = dst;
     IN_T *in = (IN_T *) src;
+#ifndef NOVOL
+    if (vol->mute) {
+        mixeng_clear (dst, samples);
+        return;
+    }
+#else
+    (void) vol;
+#endif
     while (samples--) {
-        out->l = glue(conv_,IN_T) (in[0]);
+        out->l = VOL (glue (conv_, ET) (in[0]), vol->l);
         out->r = out->l;
         out += 1;
         in += 1;
     }
 }
 
-static void glue(glue(clip_,IN_T),_from_stereo) (void *dst, const void *src,
-                                                 int samples)
+static void glue (glue (clip_, ET), _from_stereo)
+    (void *dst, const st_sample_t *src, int samples)
 {
-    st_sample_t *in = (st_sample_t *) src;
+    const st_sample_t *in = src;
     IN_T *out = (IN_T *) dst;
     while (samples--) {
-        *out++ = glue(clip_,IN_T) (in->l);
-        *out++ = glue(clip_,IN_T) (in->r);
+        *out++ = glue (clip_, ET) (in->l);
+        *out++ = glue (clip_, ET) (in->r);
         in += 1;
     }
 }
 
-static void glue(glue(clip_,IN_T),_from_mono) (void *dst, const void *src,
-                                               int samples)
+static void glue (glue (clip_, ET), _from_mono)
+    (void *dst, const st_sample_t *src, int samples)
 {
-    st_sample_t *in = (st_sample_t *) src;
+    const st_sample_t *in = src;
     IN_T *out = (IN_T *) dst;
     while (samples--) {
-        *out++ = glue(clip_,IN_T) (in->l + in->r);
+        *out++ = glue (clip_, ET) (in->l + in->r);
         in += 1;
     }
 }
 
+#undef ET
 #undef HALF
-#undef HALFT
-
+#undef VOL
diff --git a/audio/noaudio.c b/audio/noaudio.c
index a192885a72..e7936cc7b9 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -1,8 +1,8 @@
 /*
- * QEMU NULL audio output driver
- * 
- * Copyright (c) 2004 Vassili Karpov (malc)
- * 
+ * 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
@@ -23,77 +23,110 @@
  */
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "noaudio"
+#include "audio_int.h"
 
-typedef struct NoVoice {
-    HWVoice hw;
+typedef struct NoVoiceOut {
+    HWVoiceOut hw;
     int64_t old_ticks;
-} NoVoice;
+} NoVoiceOut;
 
-#define dolog(...) AUD_log ("noaudio", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+typedef struct NoVoiceIn {
+    HWVoiceIn hw;
+    int64_t old_ticks;
+} NoVoiceIn;
 
-static void no_hw_run (HWVoice *hw)
+static int no_run_out (HWVoiceOut *hw)
 {
-    NoVoice *no = (NoVoice *) hw;
-    int rpos, live, decr, samples;
-    st_sample_t *src;
+    NoVoiceOut *no = (NoVoiceOut *) hw;
+    int live, decr, samples;
     int64_t now = qemu_get_clock (vm_clock);
     int64_t ticks = now - no->old_ticks;
-    int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+    int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
 
-    if (bytes > INT_MAX)
-        samples = INT_MAX >> hw->shift;
-    else
-        samples = bytes >> hw->shift;
+    if (bytes > INT_MAX) {
+        samples = INT_MAX >> hw->info.shift;
+    }
+    else {
+        samples = bytes >> hw->info.shift;
+    }
 
-    live = pcm_hw_get_live (hw);
-    if (live <= 0)
-        return;
+    live = audio_pcm_hw_get_live_out (&no->hw);
+    if (!live) {
+        return 0;
+    }
 
     no->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);
+    hw->rpos = (hw->rpos + decr) % hw->samples;
+    return decr;
+}
 
-        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
-        memset (src, 0, convert_samples * sizeof (st_sample_t));
+static int no_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
 
-        rpos = (rpos + convert_samples) % hw->samples;
-        samples -= convert_samples;
-    }
+static int no_init_out (HWVoiceOut *hw, int freq,
+                        int nchannels, audfmt_e fmt)
+{
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
+    hw->bufsize = 4096;
+    return 0;
+}
 
-    pcm_hw_dec_live (hw, decr);
-    hw->rpos = rpos;
+static void no_fini_out (HWVoiceOut *hw)
+{
+    (void) hw;
 }
 
-static int no_hw_write (SWVoice *sw, void *buf, int len)
+static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
-    return pcm_hw_write (sw, buf, len);
+    (void) hw;
+    (void) cmd;
+    return 0;
 }
 
-static int no_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int no_init_in (HWVoiceIn *hw, int freq,
+                       int nchannels, audfmt_e fmt)
 {
-    hw->freq = freq;
-    hw->nchannels = nchannels;
-    hw->fmt = fmt;
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
     hw->bufsize = 4096;
     return 0;
 }
 
-static void no_hw_fini (HWVoice *hw)
+static void no_fini_in (HWVoiceIn *hw)
 {
     (void) hw;
 }
 
-static int no_hw_ctl (HWVoice *hw, int cmd, ...)
+static int no_run_in (HWVoiceIn *hw)
+{
+    NoVoiceIn *no = (NoVoiceIn *) hw;
+    int64_t now = qemu_get_clock (vm_clock);
+    int64_t ticks = now - no->old_ticks;
+    int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
+    int live = audio_pcm_hw_get_live_in (hw);
+    int dead = hw->samples - live;
+    int samples;
+
+    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)
+{
+    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);
+    audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
+    return to_clear;
+}
+
+static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
 {
     (void) hw;
     (void) cmd;
@@ -107,22 +140,33 @@ static void *no_audio_init (void)
 
 static void no_audio_fini (void *opaque)
 {
+    (void) opaque;
 }
 
-struct pcm_ops no_pcm_ops = {
-    no_hw_init,
-    no_hw_fini,
-    no_hw_run,
-    no_hw_write,
-    no_hw_ctl
+static struct audio_pcm_ops no_pcm_ops = {
+    no_init_out,
+    no_fini_out,
+    no_run_out,
+    no_write,
+    no_ctl_out,
+
+    no_init_in,
+    no_fini_in,
+    no_run_in,
+    no_read,
+    no_ctl_in
 };
 
-struct audio_output_driver no_output_driver = {
-    "none",
-    no_audio_init,
-    no_audio_fini,
-    &no_pcm_ops,
-    1,
-    1,
-    sizeof (NoVoice)
+struct audio_driver no_audio_driver = {
+    INIT_FIELD (name           = ) "none",
+    INIT_FIELD (descr          = ) "Timer based audio emulation",
+    INIT_FIELD (options        = ) NULL,
+    INIT_FIELD (init           = ) no_audio_init,
+    INIT_FIELD (fini           = ) no_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &no_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) INT_MAX,
+    INIT_FIELD (voice_size_out = ) sizeof (NoVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (NoVoiceIn)
 };
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 5246ebb785..ff1a034945 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -1,8 +1,8 @@
 /*
- * QEMU OSS audio output driver
- * 
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- * 
+ * 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
@@ -25,45 +25,42 @@
 #include <sys/types.h>
 #include <sys/ioctl.h>
 #include <sys/soundcard.h>
-#include <assert.h>
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "oss"
+#include "audio_int.h"
 
-typedef struct OSSVoice {
-    HWVoice hw;
+typedef struct OSSVoiceOut {
+    HWVoiceOut hw;
     void *pcm_buf;
     int fd;
     int nfrags;
     int fragsize;
     int mmapped;
     int old_optr;
-} OSSVoice;
-
-#define dolog(...) AUD_log ("oss", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+} OSSVoiceOut;
 
-#define QC_OSS_FRAGSIZE "QEMU_OSS_FRAGSIZE"
-#define QC_OSS_NFRAGS   "QEMU_OSS_NFRAGS"
-#define QC_OSS_MMAP     "QEMU_OSS_MMAP"
-#define QC_OSS_DEV      "QEMU_OSS_DEV"
-
-#define errstr() strerror (errno)
+typedef struct OSSVoiceIn {
+    HWVoiceIn hw;
+    void *pcm_buf;
+    int fd;
+    int nfrags;
+    int fragsize;
+    int old_optr;
+} OSSVoiceIn;
 
 static struct {
     int try_mmap;
     int nfrags;
     int fragsize;
-    const char *dspname;
+    const char *devpath_out;
+    const char *devpath_in;
 } conf = {
     .try_mmap = 0,
     .nfrags = 4,
     .fragsize = 4096,
-    .dspname = "/dev/dsp"
+    .devpath_out = "/dev/dsp",
+    .devpath_in = "/dev/dsp"
 };
 
 struct oss_params {
@@ -74,65 +71,141 @@ struct oss_params {
     int fragsize;
 };
 
-static int oss_hw_write (SWVoice *sw, void *buf, int len)
+static void GCC_FMT_ATTR (2, 3) oss_logerr (int err, const char *fmt, ...)
 {
-    return pcm_hw_write (sw, buf, len);
+    va_list ap;
+
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+
+    va_start (ap, fmt);
+    AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
+    va_end (ap);
 }
 
-static int AUD_to_ossfmt (audfmt_e fmt)
+static void GCC_FMT_ATTR (3, 4) oss_logerr2 (
+    int err,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Can 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 = close (*fdp);
+    if (err) {
+        oss_logerr (errno, "Failed to close file(fd=%d)\n", *fdp);
+    }
+    *fdp = -1;
+}
+
+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)
 {
     switch (fmt) {
-    case AUD_FMT_S8: return AFMT_S8;
-    case AUD_FMT_U8: return AFMT_U8;
-    case AUD_FMT_S16: return AFMT_S16_LE;
-    case AUD_FMT_U16: return AFMT_U16_LE;
+    case AUD_FMT_S8:
+        return AFMT_S8;
+
+    case AUD_FMT_U8:
+        return AFMT_U8;
+
+    case AUD_FMT_S16:
+        return AFMT_S16_LE;
+
+    case AUD_FMT_U16:
+        return AFMT_U16_LE;
+
     default:
-        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
-        exit (EXIT_FAILURE);
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+        abort ();
+#endif
+        return AFMT_U8;
     }
 }
 
-static int oss_to_audfmt (int fmt)
+static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
 {
-    switch (fmt) {
-    case AFMT_S8: return AUD_FMT_S8;
-    case AFMT_U8: return AUD_FMT_U8;
-    case AFMT_S16_LE: return AUD_FMT_S16;
-    case AFMT_U16_LE: return AUD_FMT_U16;
+    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 ("Internal logic error: Unrecognized OSS audio format %d\n"
-               "Aborting\n",
-               fmt);
-        exit (EXIT_FAILURE);
+        dolog ("Unrecognized audio format %d\n", ossfmt);
+        return -1;
     }
+
+    return 0;
 }
 
-#ifdef DEBUG_PCM
-static void oss_dump_pcm_info (struct oss_params *req, struct oss_params *obt)
+#ifdef DEBUG_MISMATCHES
+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 ("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);
+    dolog ("fragsize  |      %10d |     %10d\n",
+           req->fragsize, obt->fragsize);
 }
 #endif
 
-static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
+static int oss_open (int in, struct oss_params *req,
+                     struct oss_params *obt, int *pfd)
 {
     int fd;
     int mmmmssss;
     audio_buf_info abinfo;
     int fmt, freq, nchannels;
-    const char *dspname = conf.dspname;
+    const char *dspname = in ? conf.devpath_in : conf.devpath_out;
+    const char *typ = in ? "ADC" : "DAC";
 
-    fd = open (dspname, O_WRONLY | O_NONBLOCK);
+    fd = open (dspname, (in ? O_RDONLY : O_WRONLY) | O_NONBLOCK);
     if (-1 == fd) {
-        dolog ("Could not initialize audio hardware. Failed to open `%s':\n"
-               "Reason:%s\n",
-               dspname,
-               errstr ());
+        oss_logerr2 (errno, typ, "Failed to open `%s'\n", dspname);
         return -1;
     }
 
@@ -141,52 +214,35 @@ static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
     fmt = req->fmt;
 
     if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
-        dolog ("Could not initialize audio hardware\n"
-               "Failed to set sample size\n"
-               "Reason: %s\n",
-               errstr ());
+        oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
         goto err;
     }
 
     if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) {
-        dolog ("Could not initialize audio hardware\n"
-               "Failed to set number of channels\n"
-               "Reason: %s\n",
-               errstr ());
+        oss_logerr2 (errno, typ, "Failed to set number of channels %d\n",
+                     req->nchannels);
         goto err;
     }
 
     if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) {
-        dolog ("Could not initialize audio hardware\n"
-               "Failed to set frequency\n"
-               "Reason: %s\n",
-               errstr ());
+        oss_logerr2 (errno, typ, "Failed to set frequency %d\n", req->freq);
         goto err;
     }
 
     if (ioctl (fd, SNDCTL_DSP_NONBLOCK)) {
-        dolog ("Could not initialize audio hardware\n"
-               "Failed to set non-blocking mode\n"
-               "Reason: %s\n",
-               errstr ());
+        oss_logerr2 (errno, typ, "Failed to set non-blocking mode\n");
         goto err;
     }
 
     mmmmssss = (req->nfrags << 16) | lsbindex (req->fragsize);
     if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) {
-        dolog ("Could not initialize audio hardware\n"
-               "Failed to set buffer length (%d, %d)\n"
-               "Reason:%s\n",
-               conf.nfrags, conf.fragsize,
-               errstr ());
+        oss_logerr2 (errno, typ, "Failed to set buffer length (%d, %d)\n",
+                     req->nfrags, req->fragsize);
         goto err;
     }
 
-    if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &abinfo)) {
-        dolog ("Could not initialize audio hardware\n"
-               "Failed to get buffer length\n"
-               "Reason:%s\n",
-               errstr ());
+    if (ioctl (fd, in ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo)) {
+        oss_logerr2 (errno, typ, "Failed to get buffer length\n");
         goto err;
     }
 
@@ -202,25 +258,25 @@ static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
         (req->freq != obt->freq) ||
         (req->fragsize != obt->fragsize) ||
         (req->nfrags != obt->nfrags)) {
-#ifdef DEBUG_PCM
+#ifdef DEBUG_MISMATCHES
         dolog ("Audio parameters mismatch\n");
-        oss_dump_pcm_info (req, obt);
+        oss_dump_info (req, obt);
 #endif
     }
 
-#ifdef DEBUG_PCM
-    oss_dump_pcm_info (req, obt);
+#ifdef DEBUG
+    oss_dump_info (req, obt);
 #endif
     return 0;
 
-err:
-    close (fd);
+ err:
+    oss_anal_close (&fd);
     return -1;
 }
 
-static void oss_hw_run (HWVoice *hw)
+static int oss_run_out (HWVoiceOut *hw)
 {
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
     int err, rpos, live, decr;
     int samples;
     uint8_t *dst;
@@ -228,23 +284,25 @@ static void oss_hw_run (HWVoice *hw)
     struct audio_buf_info abinfo;
     struct count_info cntinfo;
 
-    live = pcm_hw_get_live (hw);
-    if (live <= 0)
-        return;
+    live = audio_pcm_hw_get_live_out (hw);
+    if (!live) {
+        return 0;
+    }
 
     if (oss->mmapped) {
         int bytes;
 
         err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
         if (err < 0) {
-            dolog ("SNDCTL_DSP_GETOPTR failed\nReason: %s\n", errstr ());
-            return;
+            oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
+            return 0;
         }
 
         if (cntinfo.ptr == oss->old_optr) {
-            if (abs (hw->samples - live) < 64)
-                dolog ("overrun\n");
-            return;
+            if (abs (hw->samples - live) < 64) {
+                dolog ("warning: overrun\n");
+            }
+            return 0;
         }
 
         if (cntinfo.ptr > oss->old_optr) {
@@ -254,18 +312,25 @@ static void oss_hw_run (HWVoice *hw)
             bytes = hw->bufsize + cntinfo.ptr - oss->old_optr;
         }
 
-        decr = audio_MIN (bytes >> hw->shift, live);
+        decr = audio_MIN (bytes >> hw->info.shift, live);
     }
     else {
         err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
         if (err < 0) {
-            dolog ("SNDCTL_DSP_GETOSPACE failed\nReason: %s\n", errstr ());
-            return;
+            oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
+            return 0;
+        }
+
+        if (abinfo.bytes < 0 || abinfo.bytes > hw->bufsize) {
+            ldebug ("warning: invalid available size, size=%d bufsize=%d\n",
+                    abinfo.bytes, hw->bufsize);
+            return 0;
         }
 
-        decr = audio_MIN (abinfo.bytes >> hw->shift, live);
-        if (decr <= 0)
-            return;
+        decr = audio_MIN (abinfo.bytes >> hw->info.shift, live);
+        if (!decr) {
+            return 0;
+        }
     }
 
     samples = decr;
@@ -274,33 +339,41 @@ static void oss_hw_run (HWVoice *hw)
         int left_till_end_samples = hw->samples - rpos;
         int convert_samples = audio_MIN (samples, left_till_end_samples);
 
-        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
-        dst = advance (oss->pcm_buf, rpos << hw->shift);
+        src = hw->mix_buf + rpos;
+        dst = advance (oss->pcm_buf, rpos << hw->info.shift);
 
         hw->clip (dst, src, convert_samples);
         if (!oss->mmapped) {
             int written;
 
-            written = write (oss->fd, dst, convert_samples << hw->shift);
+            written = write (oss->fd, dst, convert_samples << hw->info.shift);
             /* XXX: follow errno recommendations ? */
             if (written == -1) {
-                dolog ("Failed to write audio\nReason: %s\n", errstr ());
+                oss_logerr (
+                    errno,
+                    "Failed to write %d bytes of audio data from %p\n",
+                    convert_samples << hw->info.shift,
+                    dst
+                    );
                 continue;
             }
 
-            if (written != convert_samples << hw->shift) {
-                int wsamples = written >> hw->shift;
-                int wbytes = wsamples << hw->shift;
+            if (written != convert_samples << hw->info.shift) {
+                int wsamples = written >> hw->info.shift;
+                int wbytes = wsamples << hw->info.shift;
                 if (wbytes != written) {
-                    dolog ("Unaligned write %d, %d\n", wbytes, written);
+                    dolog ("warning: misaligned write %d (requested %d), "
+                           "alignment %d\n",
+                           wbytes, written, hw->info.align + 1);
                 }
-                memset (src, 0, wbytes);
-                decr -= samples;
+                mixeng_clear (src, wsamples);
+                decr -= wsamples;
                 rpos = (rpos + wsamples) % hw->samples;
                 break;
             }
         }
-        memset (src, 0, convert_samples * sizeof (st_sample_t));
+
+        mixeng_clear (src, convert_samples);
 
         rpos = (rpos + convert_samples) % hw->samples;
         samples -= convert_samples;
@@ -309,28 +382,24 @@ static void oss_hw_run (HWVoice *hw)
         oss->old_optr = cntinfo.ptr;
     }
 
-    pcm_hw_dec_live (hw, decr);
     hw->rpos = rpos;
+    return decr;
 }
 
-static void oss_hw_fini (HWVoice *hw)
+static void oss_fini_out (HWVoiceOut *hw)
 {
     int err;
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
 
-    ldebug ("oss_hw_fini\n");
-    err = close (oss->fd);
-    if (err) {
-        dolog ("Failed to close OSS descriptor\nReason: %s\n", errstr ());
-    }
-    oss->fd = -1;
+    ldebug ("oss_fini\n");
+    oss_anal_close (&oss->fd);
 
     if (oss->pcm_buf) {
         if (oss->mmapped) {
             err = munmap (oss->pcm_buf, hw->bufsize);
             if (err) {
-                dolog ("Failed to unmap OSS buffer\nReason: %s\n",
-                       errstr ());
+                oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
+                            oss->pcm_buf, hw->bufsize);
             }
         }
         else {
@@ -340,25 +409,38 @@ static void oss_hw_fini (HWVoice *hw)
     }
 }
 
-static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int oss_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
     struct oss_params req, obt;
+    int endianness;
+    int err;
+    int fd;
+    audfmt_e effective_fmt;
 
-    assert (!oss->fd);
-    req.fmt = AUD_to_ossfmt (fmt);
+    req.fmt = aud_to_ossfmt (fmt);
     req.freq = freq;
     req.nchannels = nchannels;
     req.fragsize = conf.fragsize;
     req.nfrags = conf.nfrags;
 
-    if (oss_open (&req, &obt, &oss->fd))
+    if (oss_open (0, &req, &obt, &fd)) {
         return -1;
+    }
 
-    hw->freq = obt.freq;
-    hw->fmt = oss_to_audfmt (obt.fmt);
-    hw->nchannels = obt.nchannels;
+    err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        oss_anal_close (&fd);
+        return -1;
+    }
 
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
     oss->nfrags = obt.nfrags;
     oss->fragsize = obt.fragsize;
     hw->bufsize = obt.nfrags * obt.fragsize;
@@ -366,22 +448,23 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     oss->mmapped = 0;
     if (conf.try_mmap) {
         oss->pcm_buf = mmap (0, hw->bufsize, PROT_READ | PROT_WRITE,
-                             MAP_SHARED, oss->fd, 0);
+                             MAP_SHARED, fd, 0);
         if (oss->pcm_buf == MAP_FAILED) {
-            dolog ("Failed to mmap OSS device\nReason: %s\n",
-                   errstr ());
+            oss_logerr (errno, "Failed to map %d bytes of DAC\n",
+                        hw->bufsize);
         } else {
             int err;
             int trig = 0;
-            if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-                dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
-                       errstr ());
+            if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
             }
             else {
                 trig = PCM_ENABLE_OUTPUT;
-                if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-                    dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
-                           "Reason: %s\n", errstr ());
+                if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                    oss_logerr (
+                        errno,
+                        "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                        );
                 }
                 else {
                     oss->mmapped = 1;
@@ -391,8 +474,8 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
             if (!oss->mmapped) {
                 err = munmap (oss->pcm_buf, hw->bufsize);
                 if (err) {
-                    dolog ("Failed to unmap OSS device\nReason: %s\n",
-                           errstr ());
+                    oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
+                                oss->pcm_buf, hw->bufsize);
                 }
             }
         }
@@ -401,31 +484,34 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     if (!oss->mmapped) {
         oss->pcm_buf = qemu_mallocz (hw->bufsize);
         if (!oss->pcm_buf) {
-            close (oss->fd);
-            oss->fd = -1;
+            oss_anal_close (&fd);
             return -1;
         }
     }
 
+    oss->fd = fd;
     return 0;
 }
 
-static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
+static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     int trig;
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
 
-    if (!oss->mmapped)
+    if (!oss->mmapped) {
         return 0;
+    }
 
     switch (cmd) {
     case VOICE_ENABLE:
         ldebug ("enabling voice\n");
-        pcm_hw_clear (hw, oss->pcm_buf, hw->samples);
+        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) {
-            dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
-                   "Reason: %s\n", errstr ());
+            oss_logerr (
+                errno,
+                "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                );
             return -1;
         }
         break;
@@ -434,8 +520,7 @@ static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
         ldebug ("disabling voice\n");
         trig = 0;
         if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-            dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
-                   errstr ());
+            oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
             return -1;
         }
         break;
@@ -443,33 +528,194 @@ static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
     return 0;
 }
 
+static int oss_init_in (HWVoiceIn *hw,
+                        int freq, int nchannels, audfmt_e fmt)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+    struct oss_params req, obt;
+    int endianness;
+    int err;
+    int fd;
+    audfmt_e effective_fmt;
+
+    req.fmt = aud_to_ossfmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.fragsize = conf.fragsize;
+    req.nfrags = conf.nfrags;
+    if (oss_open (1, &req, &obt, &fd)) {
+        return -1;
+    }
+
+    err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    oss->nfrags = obt.nfrags;
+    oss->fragsize = obt.fragsize;
+    hw->bufsize = obt.nfrags * obt.fragsize;
+    oss->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!oss->pcm_buf) {
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    oss->fd = fd;
+    return 0;
+}
+
+static void oss_fini_in (HWVoiceIn *hw)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+
+    oss_anal_close (&oss->fd);
+
+    if (oss->pcm_buf) {
+        qemu_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] = {
+        { hw->wpos, 0 },
+        { 0, 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 %d (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,
+                          &nominal_volume);
+            }
+
+            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, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
 static void *oss_audio_init (void)
 {
-    conf.fragsize = audio_get_conf_int (QC_OSS_FRAGSIZE, conf.fragsize);
-    conf.nfrags = audio_get_conf_int (QC_OSS_NFRAGS, conf.nfrags);
-    conf.try_mmap = audio_get_conf_int (QC_OSS_MMAP, conf.try_mmap);
-    conf.dspname = audio_get_conf_str (QC_OSS_DEV, conf.dspname);
     return &conf;
 }
 
 static void oss_audio_fini (void *opaque)
 {
+    (void) opaque;
 }
 
-struct pcm_ops oss_pcm_ops = {
-    oss_hw_init,
-    oss_hw_fini,
-    oss_hw_run,
-    oss_hw_write,
-    oss_hw_ctl
+static struct audio_option oss_options[] = {
+    {"FRAGSIZE", AUD_OPT_INT, &conf.fragsize,
+     "Fragment size in bytes", NULL, 0},
+    {"NFRAGS", AUD_OPT_INT, &conf.nfrags,
+     "Number of fragments", NULL, 0},
+    {"MMAP", AUD_OPT_BOOL, &conf.try_mmap,
+     "Try using memory mapped access", NULL, 0},
+    {"DAC_DEV", AUD_OPT_STR, &conf.devpath_out,
+     "Path to DAC device", NULL, 0},
+    {"ADC_DEV", AUD_OPT_STR, &conf.devpath_in,
+     "Path to ADC device", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops oss_pcm_ops = {
+    oss_init_out,
+    oss_fini_out,
+    oss_run_out,
+    oss_write,
+    oss_ctl_out,
+
+    oss_init_in,
+    oss_fini_in,
+    oss_run_in,
+    oss_read,
+    oss_ctl_in
 };
 
-struct audio_output_driver oss_output_driver = {
-    "oss",
-    oss_audio_init,
-    oss_audio_fini,
-    &oss_pcm_ops,
-    1,
-    INT_MAX,
-    sizeof (OSSVoice)
+struct audio_driver oss_audio_driver = {
+    INIT_FIELD (name           = ) "oss",
+    INIT_FIELD (descr          = ) "OSS http://www.opensound.com",
+    INIT_FIELD (options        = ) oss_options,
+    INIT_FIELD (init           = ) oss_audio_init,
+    INIT_FIELD (fini           = ) oss_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &oss_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) INT_MAX,
+    INIT_FIELD (voice_size_out = ) sizeof (OSSVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (OSSVoiceIn)
 };
diff --git a/audio/rate_template.h b/audio/rate_template.h
new file mode 100644
index 0000000000..5cc95c829a
--- /dev/null
+++ b/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, st_sample_t *ibuf, st_sample_t *obuf,
+           int *isamp, int *osamp)
+{
+    rate_t rate = (rate_t) opaque;
+    st_sample_t *istart, *iend;
+    st_sample_t *ostart, *oend;
+    st_sample_t ilast, icur, out;
+#ifdef FLOAT_MIXENG
+    real_t 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].r);
+            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) / (real_t) 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/audio/sdlaudio.c b/audio/sdlaudio.c
index 978686a071..673e2a1125 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -1,8 +1,8 @@
 /*
- * QEMU SDL audio output driver
- * 
- * Copyright (c) 2004 Vassili Karpov (malc)
- * 
+ * 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
@@ -25,22 +25,15 @@
 #include <SDL_thread.h>
 #include "vl.h"
 
-#include "audio/audio_int.h"
-
-typedef struct SDLVoice {
-    HWVoice hw;
-} SDLVoice;
-
-#define dolog(...) AUD_log ("sdl", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+#define AUDIO_CAP "sdl"
+#include "audio_int.h"
 
-#define QC_SDL_SAMPLES "QEMU_SDL_SAMPLES"
-
-#define errstr() SDL_GetError ()
+typedef struct SDLVoiceOut {
+    HWVoiceOut hw;
+    int live;
+    int rpos;
+    int decr;
+} SDLVoiceOut;
 
 static struct {
     int nb_samples;
@@ -56,91 +49,129 @@ struct SDLAudioState {
 } glob_sdl;
 typedef struct SDLAudioState SDLAudioState;
 
-static void sdl_hw_run (HWVoice *hw)
+static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
 {
-    (void) hw;
+    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)
+static int sdl_lock (SDLAudioState *s, const char *forfn)
 {
     if (SDL_LockMutex (s->mutex)) {
-        dolog ("SDL_LockMutex failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_unlock (SDLAudioState *s)
+static int sdl_unlock (SDLAudioState *s, const char *forfn)
 {
     if (SDL_UnlockMutex (s->mutex)) {
-        dolog ("SDL_UnlockMutex failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_post (SDLAudioState *s)
+static int sdl_post (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemPost (s->sem)) {
-        dolog ("SDL_SemPost failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_wait (SDLAudioState *s)
+static int sdl_wait (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemWait (s->sem)) {
-        dolog ("SDL_SemWait failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_unlock_and_post (SDLAudioState *s)
+static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
 {
-    if (sdl_unlock (s))
+    if (sdl_unlock (s, forfn)) {
         return -1;
+    }
 
-    return sdl_post (s);
-}
-
-static int sdl_hw_write (SWVoice *sw, void *buf, int len)
-{
-    int ret;
-    SDLAudioState *s = &glob_sdl;
-    sdl_lock (s);
-    ret = pcm_hw_write (sw, buf, len);
-    sdl_unlock_and_post (s);
-    return ret;
+    return sdl_post (s, forfn);
 }
 
-static int AUD_to_sdlfmt (audfmt_e fmt, int *shift)
+static int aud_to_sdlfmt (audfmt_e fmt, int *shift)
 {
-    *shift = 0;
     switch (fmt) {
-    case AUD_FMT_S8: return AUDIO_S8;
-    case AUD_FMT_U8: return AUDIO_U8;
-    case AUD_FMT_S16: *shift = 1; return AUDIO_S16LSB;
-    case AUD_FMT_U16: *shift = 1; return AUDIO_U16LSB;
+    case AUD_FMT_S8:
+        *shift = 0;
+        return AUDIO_S8;
+
+    case AUD_FMT_U8:
+        *shift = 0;
+        return AUDIO_U8;
+
+    case AUD_FMT_S16:
+        *shift = 1;
+        return AUDIO_S16LSB;
+
+    case AUD_FMT_U16:
+        *shift = 1;
+        return AUDIO_U16LSB;
+
     default:
-        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
-        exit (EXIT_FAILURE);
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+        abort ();
+#endif
+        return AUDIO_U8;
     }
 }
 
-static int sdl_to_audfmt (int fmt)
+static int sdl_to_audfmt (int sdlfmt, audfmt_e *fmt, int *endianess)
 {
-    switch (fmt) {
-    case AUDIO_S8: return AUD_FMT_S8;
-    case AUDIO_U8: return AUD_FMT_U8;
-    case AUDIO_S16LSB: return AUD_FMT_S16;
-    case AUDIO_U16LSB: return AUD_FMT_U16;
+    switch (sdlfmt) {
+    case AUDIO_S8:
+        *endianess = 0;
+        *fmt = AUD_FMT_S8;
+        break;
+
+    case AUDIO_U8:
+        *endianess = 0;
+        *fmt = AUD_FMT_U8;
+        break;
+
+    case AUDIO_S16LSB:
+        *endianess = 0;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AUDIO_U16LSB:
+        *endianess = 0;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case AUDIO_S16MSB:
+        *endianess = 1;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AUDIO_U16MSB:
+        *endianess = 1;
+        *fmt = AUD_FMT_U16;
+        break;
+
     default:
-        dolog ("Internal logic error: Unrecognized SDL audio format %d\n"
-               "Aborting\n", fmt);
-        exit (EXIT_FAILURE);
+        dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
+        return -1;
     }
+
+    return 0;
 }
 
 static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
@@ -149,7 +180,7 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
 
     status = SDL_OpenAudio (req, obt);
     if (status) {
-        dolog ("SDL_OpenAudio failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_OpenAudio failed\n");
     }
     return status;
 }
@@ -157,9 +188,9 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
 static void sdl_close (SDLAudioState *s)
 {
     if (s->initialized) {
-        sdl_lock (s);
+        sdl_lock (s, "sdl_close");
         s->exit = 1;
-        sdl_unlock_and_post (s);
+        sdl_unlock_and_post (s, "sdl_close");
         SDL_PauseAudio (1);
         SDL_CloseAudio ();
         s->initialized = 0;
@@ -168,31 +199,40 @@ static void sdl_close (SDLAudioState *s)
 
 static void sdl_callback (void *opaque, Uint8 *buf, int len)
 {
-    SDLVoice *sdl = opaque;
+    SDLVoiceOut *sdl = opaque;
     SDLAudioState *s = &glob_sdl;
-    HWVoice *hw = &sdl->hw;
-    int samples = len >> hw->shift;
+    HWVoiceOut *hw = &sdl->hw;
+    int samples = len >> hw->info.shift;
 
     if (s->exit) {
         return;
     }
 
     while (samples) {
-        int to_mix, live, decr;
+        int to_mix, decr;
 
         /* dolog ("in callback samples=%d\n", samples); */
-        sdl_wait (s);
+        sdl_wait (s, "sdl_callback");
         if (s->exit) {
             return;
         }
 
-        sdl_lock (s);
-        live = pcm_hw_get_live (hw);
-        if (live <= 0)
+        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, live);
+        to_mix = audio_MIN (samples, sdl->live);
         decr = to_mix;
         while (to_mix) {
             int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
@@ -200,44 +240,86 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
 
             /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
             hw->clip (buf, src, chunk);
-            memset (src, 0, chunk * sizeof (st_sample_t));
-            hw->rpos = (hw->rpos + chunk) % hw->samples;
+            mixeng_clear (src, chunk);
+            sdl->rpos = (sdl->rpos + chunk) % hw->samples;
             to_mix -= chunk;
-            buf += chunk << hw->shift;
+            buf += chunk << hw->info.shift;
         }
         samples -= decr;
-        pcm_hw_dec_live (hw, decr);
+        sdl->live -= decr;
+        sdl->decr += decr;
 
     again:
-        sdl_unlock (s);
+        if (sdl_unlock (s, "sdl_callback")) {
+            return;
+        }
     }
     /* dolog ("done len=%d\n", len); */
 }
 
-static void sdl_hw_fini (HWVoice *hw)
+static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
 {
-    ldebug ("sdl_hw_fini %d fixed=%d\n",
-             glob_sdl.initialized, audio_state.fixed_format);
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int sdl_run_out (HWVoiceOut *hw)
+{
+    int decr, live;
+    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+    SDLAudioState *s = &glob_sdl;
+
+    if (sdl_lock (s, "sdl_callback")) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_out (hw);
+
+    if (sdl->decr > live) {
+        ldebug ("sdl->decr %d live %d sdl->live %d\n",
+                sdl->decr,
+                live,
+                sdl->live);
+    }
+
+    decr = audio_MIN (sdl->decr, live);
+    sdl->decr -= decr;
+
+    sdl->live = live - decr;
+    hw->rpos = sdl->rpos;
+
+    if (sdl->live > 0) {
+        sdl_unlock_and_post (s, "sdl_callback");
+    }
+    else {
+        sdl_unlock (s, "sdl_callback");
+    }
+    return decr;
+}
+
+static void sdl_fini_out (HWVoiceOut *hw)
+{
+    (void) hw;
+
     sdl_close (&glob_sdl);
 }
 
-static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int sdl_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
-    SDLVoice *sdl = (SDLVoice *) hw;
+    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
     SDLAudioState *s = &glob_sdl;
     SDL_AudioSpec req, obt;
     int shift;
-
-    ldebug ("sdl_hw_init %d freq=%d fixed=%d\n",
-            s->initialized, freq, audio_state.fixed_format);
+    int endianess;
+    int err;
+    audfmt_e effective_fmt;
 
     if (nchannels != 2) {
-        dolog ("Bogus channel count %d\n", nchannels);
+        dolog ("Can not init DAC. Bogus channel count %d\n", nchannels);
         return -1;
     }
 
     req.freq = freq;
-    req.format = AUD_to_sdlfmt (fmt, &shift);
+    req.format = aud_to_sdlfmt (fmt, &shift);
     req.channels = nchannels;
     req.samples = conf.nb_samples;
     shift <<= nchannels == 2;
@@ -245,12 +327,23 @@ static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     req.callback = sdl_callback;
     req.userdata = sdl;
 
-    if (sdl_open (&req, &obt))
+    if (sdl_open (&req, &obt)) {
+        return -1;
+    }
+
+    err = sdl_to_audfmt (obt.format, &effective_fmt, &endianess);
+    if (err) {
+        sdl_close (s);
         return -1;
+    }
 
-    hw->freq = obt.freq;
-    hw->fmt = sdl_to_audfmt (obt.format);
-    hw->nchannels = obt.channels;
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.channels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianess)
+        );
     hw->bufsize = obt.samples << shift;
 
     s->initialized = 1;
@@ -259,7 +352,7 @@ static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     return 0;
 }
 
-static int sdl_hw_ctl (HWVoice *hw, int cmd, ...)
+static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     (void) hw;
 
@@ -278,24 +371,22 @@ static int sdl_hw_ctl (HWVoice *hw, int cmd, ...)
 static void *sdl_audio_init (void)
 {
     SDLAudioState *s = &glob_sdl;
-    conf.nb_samples = audio_get_conf_int (QC_SDL_SAMPLES, conf.nb_samples);
 
     if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
-        dolog ("SDL failed to initialize audio subsystem\nReason: %s\n",
-               errstr ());
+        sdl_logerr ("SDL failed to initialize audio subsystem\n");
         return NULL;
     }
 
     s->mutex = SDL_CreateMutex ();
     if (!s->mutex) {
-        dolog ("Failed to create SDL mutex\nReason: %s\n", errstr ());
+        sdl_logerr ("Failed to create SDL mutex\n");
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
     }
 
     s->sem = SDL_CreateSemaphore (0);
     if (!s->sem) {
-        dolog ("Failed to create SDL semaphore\nReason: %s\n", errstr ());
+        sdl_logerr ("Failed to create SDL semaphore\n");
         SDL_DestroyMutex (s->mutex);
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
@@ -313,20 +404,36 @@ static void sdl_audio_fini (void *opaque)
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
 }
 
-struct pcm_ops sdl_pcm_ops = {
-    sdl_hw_init,
-    sdl_hw_fini,
-    sdl_hw_run,
-    sdl_hw_write,
-    sdl_hw_ctl
+static struct audio_option sdl_options[] = {
+    {"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
+     "Size of SDL buffer in samples", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops sdl_pcm_ops = {
+    sdl_init_out,
+    sdl_fini_out,
+    sdl_run_out,
+    sdl_write_out,
+    sdl_ctl_out,
+
+    NULL,
+    NULL,
+    NULL,
+    NULL,
+    NULL
 };
 
-struct audio_output_driver sdl_output_driver = {
-    "sdl",
-    sdl_audio_init,
-    sdl_audio_fini,
-    &sdl_pcm_ops,
-    1,
-    1,
-    sizeof (SDLVoice)
+struct audio_driver sdl_audio_driver = {
+    INIT_FIELD (name           = ) "sdl",
+    INIT_FIELD (descr          = ) "SDL http://www.libsdl.org",
+    INIT_FIELD (options        = ) sdl_options,
+    INIT_FIELD (init           = ) sdl_audio_init,
+    INIT_FIELD (fini           = ) sdl_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &sdl_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) 1,
+    INIT_FIELD (max_voices_in  = ) 0,
+    INIT_FIELD (voice_size_out = ) sizeof (SDLVoiceOut),
+    INIT_FIELD (voice_size_in  = ) 0
 };
diff --git a/audio/sys-queue.h b/audio/sys-queue.h
new file mode 100644
index 0000000000..5b6e2a0a23
--- /dev/null
+++ b/audio/sys-queue.h
@@ -0,0 +1,241 @@
+/*
+ * Copyright (c) 1991, 1993
+ *	The Regents of the University of California.  All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 4. Neither the name of the University nor the names of its contributors
+ *    may be used to endorse or promote products derived from this software
+ *    without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ *	@(#)queue.h	8.3 (Berkeley) 12/13/93
+ */
+
+#ifndef	_SYS_QUEUE_H
+#define	_SYS_QUEUE_H 1
+
+/*
+ * This file defines three types of data structures: lists, tail queues,
+ * and circular queues.
+ *
+ * A list is headed by a single forward pointer (or an array of forward
+ * pointers for a hash table header). The elements are doubly linked
+ * so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list after
+ * an existing element or at the head of the list. A list may only be
+ * traversed in the forward direction.
+ *
+ * A tail queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list after
+ * an existing element, at the head of the list, or at the end of the
+ * list. A tail queue may only be traversed in the forward direction.
+ *
+ * A circle queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list before or after
+ * an existing element, at the head of the list, or at the end of the list.
+ * A circle queue may be traversed in either direction, but has a more
+ * complex end of list detection.
+ *
+ * For details on the use of these macros, see the queue(3) manual page.
+ */
+
+/*
+ * List definitions.
+ */
+#define LIST_HEAD(name, type)						\
+struct name {								\
+	struct type *lh_first;	/* first element */			\
+}
+
+#define LIST_ENTRY(type)						\
+struct {								\
+	struct type *le_next;	/* next element */			\
+	struct type **le_prev;	/* address of previous next element */	\
+}
+
+/*
+ * List functions.
+ */
+#define	LIST_INIT(head) {						\
+	(head)->lh_first = NULL;					\
+}
+
+#define LIST_INSERT_AFTER(listelm, elm, field) {			\
+	if (((elm)->field.le_next = (listelm)->field.le_next) != NULL)	\
+		(listelm)->field.le_next->field.le_prev =		\
+		    &(elm)->field.le_next;				\
+	(listelm)->field.le_next = (elm);				\
+	(elm)->field.le_prev = &(listelm)->field.le_next;		\
+}
+
+#define LIST_INSERT_HEAD(head, elm, field) {				\
+	if (((elm)->field.le_next = (head)->lh_first) != NULL)		\
+		(head)->lh_first->field.le_prev = &(elm)->field.le_next;\
+	(head)->lh_first = (elm);					\
+	(elm)->field.le_prev = &(head)->lh_first;			\
+}
+
+#define LIST_REMOVE(elm, field) {					\
+	if ((elm)->field.le_next != NULL)				\
+		(elm)->field.le_next->field.le_prev = 			\
+		    (elm)->field.le_prev;				\
+	*(elm)->field.le_prev = (elm)->field.le_next;			\
+}
+
+/*
+ * Tail queue definitions.
+ */
+#define TAILQ_HEAD(name, type)						\
+struct name {								\
+	struct type *tqh_first;	/* first element */			\
+	struct type **tqh_last;	/* addr of last next element */		\
+}
+
+#define TAILQ_ENTRY(type)						\
+struct {								\
+	struct type *tqe_next;	/* next element */			\
+	struct type **tqe_prev;	/* address of previous next element */	\
+}
+
+/*
+ * Tail queue functions.
+ */
+#define	TAILQ_INIT(head) {						\
+	(head)->tqh_first = NULL;					\
+	(head)->tqh_last = &(head)->tqh_first;				\
+}
+
+#define TAILQ_INSERT_HEAD(head, elm, field) {				\
+	if (((elm)->field.tqe_next = (head)->tqh_first) != NULL)	\
+		(elm)->field.tqe_next->field.tqe_prev =			\
+		    &(elm)->field.tqe_next;				\
+	else								\
+		(head)->tqh_last = &(elm)->field.tqe_next;		\
+	(head)->tqh_first = (elm);					\
+	(elm)->field.tqe_prev = &(head)->tqh_first;			\
+}
+
+#define TAILQ_INSERT_TAIL(head, elm, field) {				\
+	(elm)->field.tqe_next = NULL;					\
+	(elm)->field.tqe_prev = (head)->tqh_last;			\
+	*(head)->tqh_last = (elm);					\
+	(head)->tqh_last = &(elm)->field.tqe_next;			\
+}
+
+#define TAILQ_INSERT_AFTER(head, listelm, elm, field) {			\
+	if (((elm)->field.tqe_next = (listelm)->field.tqe_next) != NULL)\
+		(elm)->field.tqe_next->field.tqe_prev = 		\
+		    &(elm)->field.tqe_next;				\
+	else								\
+		(head)->tqh_last = &(elm)->field.tqe_next;		\
+	(listelm)->field.tqe_next = (elm);				\
+	(elm)->field.tqe_prev = &(listelm)->field.tqe_next;		\
+}
+
+#define TAILQ_REMOVE(head, elm, field) {				\
+	if (((elm)->field.tqe_next) != NULL)				\
+		(elm)->field.tqe_next->field.tqe_prev = 		\
+		    (elm)->field.tqe_prev;				\
+	else								\
+		(head)->tqh_last = (elm)->field.tqe_prev;		\
+	*(elm)->field.tqe_prev = (elm)->field.tqe_next;			\
+}
+
+/*
+ * Circular queue definitions.
+ */
+#define CIRCLEQ_HEAD(name, type)					\
+struct name {								\
+	struct type *cqh_first;		/* first element */		\
+	struct type *cqh_last;		/* last element */		\
+}
+
+#define CIRCLEQ_ENTRY(type)						\
+struct {								\
+	struct type *cqe_next;		/* next element */		\
+	struct type *cqe_prev;		/* previous element */		\
+}
+
+/*
+ * Circular queue functions.
+ */
+#define	CIRCLEQ_INIT(head) {						\
+	(head)->cqh_first = (void *)(head);				\
+	(head)->cqh_last = (void *)(head);				\
+}
+
+#define CIRCLEQ_INSERT_AFTER(head, listelm, elm, field) {		\
+	(elm)->field.cqe_next = (listelm)->field.cqe_next;		\
+	(elm)->field.cqe_prev = (listelm);				\
+	if ((listelm)->field.cqe_next == (void *)(head))		\
+		(head)->cqh_last = (elm);				\
+	else								\
+		(listelm)->field.cqe_next->field.cqe_prev = (elm);	\
+	(listelm)->field.cqe_next = (elm);				\
+}
+
+#define CIRCLEQ_INSERT_BEFORE(head, listelm, elm, field) {		\
+	(elm)->field.cqe_next = (listelm);				\
+	(elm)->field.cqe_prev = (listelm)->field.cqe_prev;		\
+	if ((listelm)->field.cqe_prev == (void *)(head))		\
+		(head)->cqh_first = (elm);				\
+	else								\
+		(listelm)->field.cqe_prev->field.cqe_next = (elm);	\
+	(listelm)->field.cqe_prev = (elm);				\
+}
+
+#define CIRCLEQ_INSERT_HEAD(head, elm, field) {				\
+	(elm)->field.cqe_next = (head)->cqh_first;			\
+	(elm)->field.cqe_prev = (void *)(head);				\
+	if ((head)->cqh_last == (void *)(head))				\
+		(head)->cqh_last = (elm);				\
+	else								\
+		(head)->cqh_first->field.cqe_prev = (elm);		\
+	(head)->cqh_first = (elm);					\
+}
+
+#define CIRCLEQ_INSERT_TAIL(head, elm, field) {				\
+	(elm)->field.cqe_next = (void *)(head);				\
+	(elm)->field.cqe_prev = (head)->cqh_last;			\
+	if ((head)->cqh_first == (void *)(head))			\
+		(head)->cqh_first = (elm);				\
+	else								\
+		(head)->cqh_last->field.cqe_next = (elm);		\
+	(head)->cqh_last = (elm);					\
+}
+
+#define	CIRCLEQ_REMOVE(head, elm, field) {				\
+	if ((elm)->field.cqe_next == (void *)(head))			\
+		(head)->cqh_last = (elm)->field.cqe_prev;		\
+	else								\
+		(elm)->field.cqe_next->field.cqe_prev =			\
+		    (elm)->field.cqe_prev;				\
+	if ((elm)->field.cqe_prev == (void *)(head))			\
+		(head)->cqh_first = (elm)->field.cqe_next;		\
+	else								\
+		(elm)->field.cqe_prev->field.cqe_next =			\
+		    (elm)->field.cqe_next;				\
+}
+#endif	/* sys/queue.h */
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 5680161c72..e9bd87872f 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -1,8 +1,8 @@
 /*
- * QEMU WAV audio output driver
- * 
- * Copyright (c) 2004 Vassili Karpov (malc)
- * 
+ * 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
@@ -23,22 +23,16 @@
  */
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "wav"
+#include "audio_int.h"
 
-typedef struct WAVVoice {
-    HWVoice hw;
+typedef struct WAVVoiceOut {
+    HWVoiceOut hw;
     QEMUFile *f;
     int64_t old_ticks;
     void *pcm_buf;
     int total_samples;
-} WAVVoice;
-
-#define dolog(...) AUD_log ("wav", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+} WAVVoiceOut;
 
 static struct {
     const char *wav_path;
@@ -46,24 +40,27 @@ static struct {
     .wav_path = "qemu.wav"
 };
 
-static void wav_hw_run (HWVoice *hw)
+static int wav_run_out (HWVoiceOut *hw)
 {
-    WAVVoice *wav = (WAVVoice *) hw;
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     int rpos, live, decr, samples;
     uint8_t *dst;
     st_sample_t *src;
     int64_t now = qemu_get_clock (vm_clock);
     int64_t ticks = now - wav->old_ticks;
-    int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+    int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
 
-    if (bytes > INT_MAX)
-        samples = INT_MAX >> hw->shift;
-    else
-        samples = bytes >> hw->shift;
+    if (bytes > INT_MAX) {
+        samples = INT_MAX >> hw->info.shift;
+    }
+    else {
+        samples = bytes >> hw->info.shift;
+    }
 
-    live = pcm_hw_get_live (hw);
-    if (live <= 0)
-        return;
+    live = audio_pcm_hw_get_live_out (hw);
+    if (!live) {
+        return 0;
+    }
 
     wav->old_ticks = now;
     decr = audio_MIN (live, samples);
@@ -73,25 +70,25 @@ static void wav_hw_run (HWVoice *hw)
         int left_till_end_samples = hw->samples - rpos;
         int convert_samples = audio_MIN (samples, left_till_end_samples);
 
-        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
-        dst = advance (wav->pcm_buf, rpos << hw->shift);
+        src = hw->mix_buf + rpos;
+        dst = advance (wav->pcm_buf, rpos << hw->info.shift);
 
         hw->clip (dst, src, convert_samples);
-        qemu_put_buffer (wav->f, dst, convert_samples << hw->shift);
-        memset (src, 0, convert_samples * sizeof (st_sample_t));
+        qemu_put_buffer (wav->f, dst, convert_samples << hw->info.shift);
+        mixeng_clear (src, convert_samples);
 
         rpos = (rpos + convert_samples) % hw->samples;
         samples -= convert_samples;
         wav->total_samples += convert_samples;
     }
 
-    pcm_hw_dec_live (hw, decr);
     hw->rpos = rpos;
+    return decr;
 }
 
-static int wav_hw_write (SWVoice *sw, void *buf, int len)
+static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
 {
-    return pcm_hw_write (sw, buf, len);
+    return audio_pcm_sw_write (sw, buf, len);
 }
 
 /* VICE code: Store number as little endian. */
@@ -104,10 +101,10 @@ static void le_store (uint8_t *buf, uint32_t val, int len)
     }
 }
 
-static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int wav_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
-    WAVVoice *wav = (WAVVoice *) hw;
-    int bits16 = 0, stereo = audio_state.fixed_channels == 2;
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+    int bits16;
     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,
@@ -115,34 +112,50 @@ static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
         0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
     };
 
-    switch (audio_state.fixed_fmt) {
+    freq = audio_state.fixed_freq_out;
+    fmt = audio_state.fixed_fmt_out;
+    nchannels = audio_state.fixed_channels_out;
+
+    switch (fmt) {
     case AUD_FMT_S8:
     case AUD_FMT_U8:
+        bits16 = 0;
         break;
 
     case AUD_FMT_S16:
     case AUD_FMT_U16:
         bits16 = 1;
         break;
+
+    default:
+        dolog ("Internal logic error bad format %d\n", fmt);
+        return -1;
     }
 
     hdr[34] = bits16 ? 0x10 : 0x08;
-    hw->freq = 44100;
-    hw->nchannels = stereo ? 2 : 1;
-    hw->fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+    audio_pcm_init_info (
+        &hw->info,
+        freq,
+        nchannels,
+        bits16 ? AUD_FMT_S16 : AUD_FMT_U8,
+        audio_need_to_swap_endian (0)
+        );
     hw->bufsize = 4096;
     wav->pcm_buf = qemu_mallocz (hw->bufsize);
-    if (!wav->pcm_buf)
+    if (!wav->pcm_buf) {
+        dolog ("Can not initialize WAV buffer of %d bytes\n",
+               hw->bufsize);
         return -1;
+    }
 
-    le_store (hdr + 22, hw->nchannels, 2);
-    le_store (hdr + 24, hw->freq, 4);
-    le_store (hdr + 28, hw->freq << (bits16 + stereo), 4);
-    le_store (hdr + 32, 1 << (bits16 + stereo), 2);
+    le_store (hdr + 22, hw->info.nchannels, 2);
+    le_store (hdr + 24, hw->info.freq, 4);
+    le_store (hdr + 28, hw->info.freq << (bits16 + (nchannels == 2)), 4);
+    le_store (hdr + 32, 1 << (bits16 + (nchannels == 2)), 2);
 
     wav->f = fopen (conf.wav_path, "wb");
     if (!wav->f) {
-        dolog ("failed to open wave file `%s'\nReason: %s\n",
+        dolog ("Failed to open wave file `%s'\nReason: %s\n",
                conf.wav_path, strerror (errno));
         qemu_free (wav->pcm_buf);
         wav->pcm_buf = NULL;
@@ -153,17 +166,18 @@ static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     return 0;
 }
 
-static void wav_hw_fini (HWVoice *hw)
+static void wav_fini_out (HWVoiceOut *hw)
 {
-    WAVVoice *wav = (WAVVoice *) hw;
-    int stereo = hw->nchannels == 2;
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+    int stereo = hw->info.nchannels == 2;
     uint8_t rlen[4];
     uint8_t dlen[4];
     uint32_t rifflen = (wav->total_samples << stereo) + 36;
     uint32_t datalen = wav->total_samples << stereo;
 
-    if (!wav->f || !hw->active)
+    if (!wav->f || !hw->active) {
         return;
+    }
 
     le_store (rlen, rifflen, 4);
     le_store (dlen, datalen, 4);
@@ -181,7 +195,7 @@ static void wav_hw_fini (HWVoice *hw)
     wav->pcm_buf = NULL;
 }
 
-static int wav_hw_ctl (HWVoice *hw, int cmd, ...)
+static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     (void) hw;
     (void) cmd;
@@ -195,23 +209,41 @@ static void *wav_audio_init (void)
 
 static void wav_audio_fini (void *opaque)
 {
+    (void) opaque;
     ldebug ("wav_fini");
 }
 
-struct pcm_ops wav_pcm_ops = {
-    wav_hw_init,
-    wav_hw_fini,
-    wav_hw_run,
-    wav_hw_write,
-    wav_hw_ctl
+struct audio_option wav_options[] = {
+    {"PATH", AUD_OPT_STR, &conf.wav_path,
+     "Path to wave file", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+struct audio_pcm_ops wav_pcm_ops = {
+    wav_init_out,
+    wav_fini_out,
+    wav_run_out,
+    wav_write_out,
+    wav_ctl_out,
+
+    NULL,
+    NULL,
+    NULL,
+    NULL,
+    NULL
 };
 
-struct audio_output_driver wav_output_driver = {
-    "wav",
-    wav_audio_init,
-    wav_audio_fini,
-    &wav_pcm_ops,
-    1,
-    1,
-    sizeof (WAVVoice)
+struct audio_driver wav_audio_driver = {
+    INIT_FIELD (name           = ) "wav",
+    INIT_FIELD (descr          = )
+    "WAV renderer http://wikipedia.org/wiki/WAV",
+    INIT_FIELD (options        = ) wav_options,
+    INIT_FIELD (init           = ) wav_audio_init,
+    INIT_FIELD (fini           = ) wav_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &wav_pcm_ops,
+    INIT_FIELD (can_be_default = ) 0,
+    INIT_FIELD (max_voices_out = ) 1,
+    INIT_FIELD (max_voices_in  = ) 0,
+    INIT_FIELD (voice_size_out = ) sizeof (WAVVoiceOut),
+    INIT_FIELD (voice_size_in  = ) 0
 };