Introduce Virtio-loopback epsilon release:

Epsilon release introduces a new compatibility layer which make virtio-loopback
design to work with QEMU and rust-vmm vhost-user backend without require any
changes.

Signed-off-by: Timos Ampelikiotis <t.ampelikiotis@virtualopensystems.com>
Change-Id: I52e57563e08a7d0bdc002f8e928ee61ba0c53dd9

1 files changed, 2222 insertions, 0 deletions

diff --git a/audio/audio.c b/audio/audio.c
new file mode 100644
index 000000000..54a153c0e
--- /dev/null
+++ b/audio/audio.c
@@ -0,0 +1,2222 @@
+/*
+ * QEMU Audio subsystem
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "qemu/osdep.h"
+#include "audio.h"
+#include "migration/vmstate.h"
+#include "monitor/monitor.h"
+#include "qemu/timer.h"
+#include "qapi/error.h"
+#include "qapi/qobject-input-visitor.h"
+#include "qapi/qapi-visit-audio.h"
+#include "qemu/cutils.h"
+#include "qemu/module.h"
+#include "qemu-common.h"
+#include "sysemu/replay.h"
+#include "sysemu/runstate.h"
+#include "ui/qemu-spice.h"
+#include "trace.h"
+
+#define AUDIO_CAP "audio"
+#include "audio_int.h"
+
+/* #define DEBUG_LIVE */
+/* #define DEBUG_OUT */
+/* #define DEBUG_CAPTURE */
+/* #define DEBUG_POLL */
+
+#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
+
+
+/* Order of CONFIG_AUDIO_DRIVERS is import.
+   The 1st one is the one used by default, that is the reason
+    that we generate the list.
+*/
+const char *audio_prio_list[] = {
+    "spice",
+    CONFIG_AUDIO_DRIVERS
+    "none",
+    "wav",
+    NULL
+};
+
+static QLIST_HEAD(, audio_driver) audio_drivers;
+static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
+
+void audio_driver_register(audio_driver *drv)
+{
+    QLIST_INSERT_HEAD(&audio_drivers, drv, next);
+}
+
+audio_driver *audio_driver_lookup(const char *name)
+{
+    struct audio_driver *d;
+
+    QLIST_FOREACH(d, &audio_drivers, next) {
+        if (strcmp(name, d->name) == 0) {
+            return d;
+        }
+    }
+
+    audio_module_load_one(name);
+    QLIST_FOREACH(d, &audio_drivers, next) {
+        if (strcmp(name, d->name) == 0) {
+            return d;
+        }
+    }
+
+    return NULL;
+}
+
+static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
+    QTAILQ_HEAD_INITIALIZER(audio_states);
+
+const struct mixeng_volume nominal_volume = {
+    .mute = 0,
+#ifdef FLOAT_MIXENG
+    .r = 1.0,
+    .l = 1.0,
+#else
+    .r = 1ULL << 32,
+    .l = 1ULL << 32,
+#endif
+};
+
+static bool legacy_config = true;
+
+int audio_bug (const char *funcname, int cond)
+{
+    if (cond) {
+        static int shown;
+
+        AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
+        if (!shown) {
+            shown = 1;
+            AUD_log (NULL, "Save all your work and restart without audio\n");
+            AUD_log (NULL, "I am sorry\n");
+        }
+        AUD_log (NULL, "Context:\n");
+        abort();
+    }
+
+    return cond;
+}
+
+static inline int audio_bits_to_index (int bits)
+{
+    switch (bits) {
+    case 8:
+        return 0;
+
+    case 16:
+        return 1;
+
+    case 32:
+        return 2;
+
+    default:
+        audio_bug ("bits_to_index", 1);
+        AUD_log (NULL, "invalid bits %d\n", bits);
+        return 0;
+    }
+}
+
+void *audio_calloc (const char *funcname, int nmemb, size_t size)
+{
+    int cond;
+    size_t len;
+
+    len = nmemb * size;
+    cond = !nmemb || !size;
+    cond |= nmemb < 0;
+    cond |= len < size;
+
+    if (audio_bug ("audio_calloc", cond)) {
+        AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
+                 funcname);
+        AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
+        return NULL;
+    }
+
+    return g_malloc0 (len);
+}
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap)
+{
+    if (cap) {
+        fprintf(stderr, "%s: ", cap);
+    }
+
+    vfprintf(stderr, fmt, ap);
+}
+
+void AUD_log (const char *cap, const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (cap, fmt, ap);
+    va_end (ap);
+}
+
+static void audio_print_settings (struct audsettings *as)
+{
+    dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
+
+    switch (as->fmt) {
+    case AUDIO_FORMAT_S8:
+        AUD_log (NULL, "S8");
+        break;
+    case AUDIO_FORMAT_U8:
+        AUD_log (NULL, "U8");
+        break;
+    case AUDIO_FORMAT_S16:
+        AUD_log (NULL, "S16");
+        break;
+    case AUDIO_FORMAT_U16:
+        AUD_log (NULL, "U16");
+        break;
+    case AUDIO_FORMAT_S32:
+        AUD_log (NULL, "S32");
+        break;
+    case AUDIO_FORMAT_U32:
+        AUD_log (NULL, "U32");
+        break;
+    case AUDIO_FORMAT_F32:
+        AUD_log (NULL, "F32");
+        break;
+    default:
+        AUD_log (NULL, "invalid(%d)", as->fmt);
+        break;
+    }
+
+    AUD_log (NULL, " endianness=");
+    switch (as->endianness) {
+    case 0:
+        AUD_log (NULL, "little");
+        break;
+    case 1:
+        AUD_log (NULL, "big");
+        break;
+    default:
+        AUD_log (NULL, "invalid");
+        break;
+    }
+    AUD_log (NULL, "\n");
+}
+
+static int audio_validate_settings (struct audsettings *as)
+{
+    int invalid;
+
+    invalid = as->nchannels < 1;
+    invalid |= as->endianness != 0 && as->endianness != 1;
+
+    switch (as->fmt) {
+    case AUDIO_FORMAT_S8:
+    case AUDIO_FORMAT_U8:
+    case AUDIO_FORMAT_S16:
+    case AUDIO_FORMAT_U16:
+    case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_U32:
+    case AUDIO_FORMAT_F32:
+        break;
+    default:
+        invalid = 1;
+        break;
+    }
+
+    invalid |= as->freq <= 0;
+    return invalid ? -1 : 0;
+}
+
+static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
+{
+    int bits = 8;
+    bool is_signed = false, is_float = false;
+
+    switch (as->fmt) {
+    case AUDIO_FORMAT_S8:
+        is_signed = true;
+        /* fall through */
+    case AUDIO_FORMAT_U8:
+        break;
+
+    case AUDIO_FORMAT_S16:
+        is_signed = true;
+        /* fall through */
+    case AUDIO_FORMAT_U16:
+        bits = 16;
+        break;
+
+    case AUDIO_FORMAT_F32:
+        is_float = true;
+        /* fall through */
+    case AUDIO_FORMAT_S32:
+        is_signed = true;
+        /* fall through */
+    case AUDIO_FORMAT_U32:
+        bits = 32;
+        break;
+
+    default:
+        abort();
+    }
+    return info->freq == as->freq
+        && info->nchannels == as->nchannels
+        && info->is_signed == is_signed
+        && info->is_float == is_float
+        && info->bits == bits
+        && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
+}
+
+void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
+{
+    int bits = 8, mul;
+    bool is_signed = false, is_float = false;
+
+    switch (as->fmt) {
+    case AUDIO_FORMAT_S8:
+        is_signed = true;
+        /* fall through */
+    case AUDIO_FORMAT_U8:
+        mul = 1;
+        break;
+
+    case AUDIO_FORMAT_S16:
+        is_signed = true;
+        /* fall through */
+    case AUDIO_FORMAT_U16:
+        bits = 16;
+        mul = 2;
+        break;
+
+    case AUDIO_FORMAT_F32:
+        is_float = true;
+        /* fall through */
+    case AUDIO_FORMAT_S32:
+        is_signed = true;
+        /* fall through */
+    case AUDIO_FORMAT_U32:
+        bits = 32;
+        mul = 4;
+        break;
+
+    default:
+        abort();
+    }
+
+    info->freq = as->freq;
+    info->bits = bits;
+    info->is_signed = is_signed;
+    info->is_float = is_float;
+    info->nchannels = as->nchannels;
+    info->bytes_per_frame = as->nchannels * mul;
+    info->bytes_per_second = info->freq * info->bytes_per_frame;
+    info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
+}
+
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
+{
+    if (!len) {
+        return;
+    }
+
+    if (info->is_signed || info->is_float) {
+        memset(buf, 0x00, len * info->bytes_per_frame);
+    } else {
+        switch (info->bits) {
+        case 8:
+            memset(buf, 0x80, len * info->bytes_per_frame);
+            break;
+
+        case 16:
+            {
+                int i;
+                uint16_t *p = buf;
+                short s = INT16_MAX;
+
+                if (info->swap_endianness) {
+                    s = bswap16 (s);
+                }
+
+                for (i = 0; i < len * info->nchannels; i++) {
+                    p[i] = s;
+                }
+            }
+            break;
+
+        case 32:
+            {
+                int i;
+                uint32_t *p = buf;
+                int32_t s = INT32_MAX;
+
+                if (info->swap_endianness) {
+                    s = bswap32 (s);
+                }
+
+                for (i = 0; i < len * info->nchannels; i++) {
+                    p[i] = s;
+                }
+            }
+            break;
+
+        default:
+            AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
+                     info->bits);
+            break;
+        }
+    }
+}
+
+/*
+ * Capture
+ */
+static void noop_conv (struct st_sample *dst, const void *src, int samples)
+{
+    (void) src;
+    (void) dst;
+    (void) samples;
+}
+
+static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
+                                                        struct audsettings *as)
+{
+    CaptureVoiceOut *cap;
+
+    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+        if (audio_pcm_info_eq (&cap->hw.info, as)) {
+            return cap;
+        }
+    }
+    return NULL;
+}
+
+static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
+{
+    struct capture_callback *cb;
+
+#ifdef DEBUG_CAPTURE
+    dolog ("notification %d sent\n", cmd);
+#endif
+    for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+        cb->ops.notify (cb->opaque, cmd);
+    }
+}
+
+static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
+{
+    if (cap->hw.enabled != enabled) {
+        audcnotification_e cmd;
+        cap->hw.enabled = enabled;
+        cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
+        audio_notify_capture (cap, cmd);
+    }
+}
+
+static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
+{
+    HWVoiceOut *hw = &cap->hw;
+    SWVoiceOut *sw;
+    int enabled = 0;
+
+    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        if (sw->active) {
+            enabled = 1;
+            break;
+        }
+    }
+    audio_capture_maybe_changed (cap, enabled);
+}
+
+static void audio_detach_capture (HWVoiceOut *hw)
+{
+    SWVoiceCap *sc = hw->cap_head.lh_first;
+
+    while (sc) {
+        SWVoiceCap *sc1 = sc->entries.le_next;
+        SWVoiceOut *sw = &sc->sw;
+        CaptureVoiceOut *cap = sc->cap;
+        int was_active = sw->active;
+
+        if (sw->rate) {
+            st_rate_stop (sw->rate);
+            sw->rate = NULL;
+        }
+
+        QLIST_REMOVE (sw, entries);
+        QLIST_REMOVE (sc, entries);
+        g_free (sc);
+        if (was_active) {
+            /* We have removed soft voice from the capture:
+               this might have changed the overall status of the capture
+               since this might have been the only active voice */
+            audio_recalc_and_notify_capture (cap);
+        }
+        sc = sc1;
+    }
+}
+
+static int audio_attach_capture (HWVoiceOut *hw)
+{
+    AudioState *s = hw->s;
+    CaptureVoiceOut *cap;
+
+    audio_detach_capture (hw);
+    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+        SWVoiceCap *sc;
+        SWVoiceOut *sw;
+        HWVoiceOut *hw_cap = &cap->hw;
+
+        sc = g_malloc0(sizeof(*sc));
+
+        sc->cap = cap;
+        sw = &sc->sw;
+        sw->hw = hw_cap;
+        sw->info = hw->info;
+        sw->empty = 1;
+        sw->active = hw->enabled;
+        sw->conv = noop_conv;
+        sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
+        sw->vol = nominal_volume;
+        sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
+        if (!sw->rate) {
+            dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
+            g_free (sw);
+            return -1;
+        }
+        QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
+        QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
+#ifdef DEBUG_CAPTURE
+        sw->name = g_strdup_printf ("for %p %d,%d,%d",
+                                    hw, sw->info.freq, sw->info.bits,
+                                    sw->info.nchannels);
+        dolog ("Added %s active = %d\n", sw->name, sw->active);
+#endif
+        if (sw->active) {
+            audio_capture_maybe_changed (cap, 1);
+        }
+    }
+    return 0;
+}
+
+/*
+ * Hard voice (capture)
+ */
+static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
+{
+    SWVoiceIn *sw;
+    size_t m = hw->total_samples_captured;
+
+    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        if (sw->active) {
+            m = MIN (m, sw->total_hw_samples_acquired);
+        }
+    }
+    return m;
+}
+
+static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
+{
+    size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
+    if (audio_bug(__func__, live > hw->conv_buf->size)) {
+        dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+        return 0;
+    }
+    return live;
+}
+
+static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
+{
+    size_t clipped = 0;
+    size_t pos = hw->mix_buf->pos;
+
+    while (len) {
+        st_sample *src = hw->mix_buf->samples + pos;
+        uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
+        size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
+        size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
+
+        hw->clip(dst, src, samples_to_clip);
+
+        pos = (pos + samples_to_clip) % hw->mix_buf->size;
+        len -= samples_to_clip;
+        clipped += samples_to_clip;
+    }
+}
+
+/*
+ * Soft voice (capture)
+ */
+static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
+{
+    HWVoiceIn *hw = sw->hw;
+    ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+    ssize_t rpos;
+
+    if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
+        dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+        return 0;
+    }
+
+    rpos = hw->conv_buf->pos - live;
+    if (rpos >= 0) {
+        return rpos;
+    } else {
+        return hw->conv_buf->size + rpos;
+    }
+}
+
+static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
+{
+    HWVoiceIn *hw = sw->hw;
+    size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+    struct st_sample *src, *dst = sw->buf;
+
+    rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
+
+    live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+    if (audio_bug(__func__, live > hw->conv_buf->size)) {
+        dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+        return 0;
+    }
+
+    samples = size / sw->info.bytes_per_frame;
+    if (!live) {
+        return 0;
+    }
+
+    swlim = (live * sw->ratio) >> 32;
+    swlim = MIN (swlim, samples);
+
+    while (swlim) {
+        src = hw->conv_buf->samples + rpos;
+        if (hw->conv_buf->pos > rpos) {
+            isamp = hw->conv_buf->pos - rpos;
+        } else {
+            isamp = hw->conv_buf->size - rpos;
+        }
+
+        if (!isamp) {
+            break;
+        }
+        osamp = swlim;
+
+        st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
+        swlim -= osamp;
+        rpos = (rpos + isamp) % hw->conv_buf->size;
+        dst += osamp;
+        ret += osamp;
+        total += isamp;
+    }
+
+    if (hw->pcm_ops && !hw->pcm_ops->volume_in) {
+        mixeng_volume (sw->buf, ret, &sw->vol);
+    }
+
+    sw->clip (buf, sw->buf, ret);
+    sw->total_hw_samples_acquired += total;
+    return ret * sw->info.bytes_per_frame;
+}
+
+/*
+ * Hard voice (playback)
+ */
+static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
+{
+    SWVoiceOut *sw;
+    size_t m = SIZE_MAX;
+    int nb_live = 0;
+
+    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+        if (sw->active || !sw->empty) {
+            m = MIN (m, sw->total_hw_samples_mixed);
+            nb_live += 1;
+        }
+    }
+
+    *nb_livep = nb_live;
+    return m;
+}
+
+static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
+{
+    size_t smin;
+    int nb_live1;
+
+    smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
+    if (nb_live) {
+        *nb_live = nb_live1;
+    }
+
+    if (nb_live1) {
+        size_t live = smin;
+
+        if (audio_bug(__func__, live > hw->mix_buf->size)) {
+            dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
+            return 0;
+        }
+        return live;
+    }
+    return 0;
+}
+
+/*
+ * Soft voice (playback)
+ */
+static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
+{
+    size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+    size_t ret = 0, pos = 0, total = 0;
+
+    if (!sw) {
+        return size;
+    }
+
+    hwsamples = sw->hw->mix_buf->size;
+
+    live = sw->total_hw_samples_mixed;
+    if (audio_bug(__func__, live > hwsamples)) {
+        dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
+        return 0;
+    }
+
+    if (live == hwsamples) {
+#ifdef DEBUG_OUT
+        dolog ("%s is full %zu\n", sw->name, live);
+#endif
+        return 0;
+    }
+
+    wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
+    samples = size / sw->info.bytes_per_frame;
+
+    dead = hwsamples - live;
+    swlim = ((int64_t) dead << 32) / sw->ratio;
+    swlim = MIN (swlim, samples);
+    if (swlim) {
+        sw->conv (sw->buf, buf, swlim);
+
+        if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) {
+            mixeng_volume (sw->buf, swlim, &sw->vol);
+        }
+    }
+
+    while (swlim) {
+        dead = hwsamples - live;
+        left = hwsamples - wpos;
+        blck = MIN (dead, left);
+        if (!blck) {
+            break;
+        }
+        isamp = swlim;
+        osamp = blck;
+        st_rate_flow_mix (
+            sw->rate,
+            sw->buf + pos,
+            sw->hw->mix_buf->samples + wpos,
+            &isamp,
+            &osamp
+            );
+        ret += isamp;
+        swlim -= isamp;
+        pos += isamp;
+        live += osamp;
+        wpos = (wpos + osamp) % hwsamples;
+        total += osamp;
+    }
+
+    sw->total_hw_samples_mixed += total;
+    sw->empty = sw->total_hw_samples_mixed == 0;
+
+#ifdef DEBUG_OUT
+    dolog (
+        "%s: write size %zu ret %zu total sw %zu\n",
+        SW_NAME (sw),
+        size / sw->info.bytes_per_frame,
+        ret,
+        sw->total_hw_samples_mixed
+        );
+#endif
+
+    return ret * sw->info.bytes_per_frame;
+}
+
+#ifdef DEBUG_AUDIO
+static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
+{
+    dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
+          cap, info->bits, info->is_signed, info->is_float, info->freq,
+          info->nchannels);
+}
+#endif
+
+#define DAC
+#include "audio_template.h"
+#undef DAC
+#include "audio_template.h"
+
+/*
+ * Timer
+ */
+static int audio_is_timer_needed(AudioState *s)
+{
+    HWVoiceIn *hwi = NULL;
+    HWVoiceOut *hwo = NULL;
+
+    while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
+        if (!hwo->poll_mode) {
+            return 1;
+        }
+    }
+    while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
+        if (!hwi->poll_mode) {
+            return 1;
+        }
+    }
+    return 0;
+}
+
+static void audio_reset_timer (AudioState *s)
+{
+    if (audio_is_timer_needed(s)) {
+        timer_mod_anticipate_ns(s->ts,
+            qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
+        if (!s->timer_running) {
+            s->timer_running = true;
+            s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+            trace_audio_timer_start(s->period_ticks / SCALE_MS);
+        }
+    } else {
+        timer_del(s->ts);
+        if (s->timer_running) {
+            s->timer_running = false;
+            trace_audio_timer_stop();
+        }
+    }
+}
+
+static void audio_timer (void *opaque)
+{
+    int64_t now, diff;
+    AudioState *s = opaque;
+
+    now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+    diff = now - s->timer_last;
+    if (diff > s->period_ticks * 3 / 2) {
+        trace_audio_timer_delayed(diff / SCALE_MS);
+    }
+    s->timer_last = now;
+
+    audio_run(s, "timer");
+    audio_reset_timer(s);
+}
+
+/*
+ * Public API
+ */
+size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
+{
+    HWVoiceOut *hw;
+
+    if (!sw) {
+        /* XXX: Consider options */
+        return size;
+    }
+    hw = sw->hw;
+
+    if (!hw->enabled) {
+        dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
+        return 0;
+    }
+
+    if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
+        return audio_pcm_sw_write(sw, buf, size);
+    } else {
+        return hw->pcm_ops->write(hw, buf, size);
+    }
+}
+
+size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
+{
+    HWVoiceIn *hw;
+
+    if (!sw) {
+        /* XXX: Consider options */
+        return size;
+    }
+    hw = sw->hw;
+
+    if (!hw->enabled) {
+        dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
+        return 0;
+    }
+
+    if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
+        return audio_pcm_sw_read(sw, buf, size);
+    } else {
+        return hw->pcm_ops->read(hw, buf, size);
+    }
+}
+
+int AUD_get_buffer_size_out(SWVoiceOut *sw)
+{
+    return sw->hw->samples * sw->hw->info.bytes_per_frame;
+}
+
+void AUD_set_active_out (SWVoiceOut *sw, int on)
+{
+    HWVoiceOut *hw;
+
+    if (!sw) {
+        return;
+    }
+
+    hw = sw->hw;
+    if (sw->active != on) {
+        AudioState *s = sw->s;
+        SWVoiceOut *temp_sw;
+        SWVoiceCap *sc;
+
+        if (on) {
+            hw->pending_disable = 0;
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                if (s->vm_running) {
+                    if (hw->pcm_ops->enable_out) {
+                        hw->pcm_ops->enable_out(hw, true);
+                    }
+                    audio_reset_timer (s);
+                }
+            }
+        } else {
+            if (hw->enabled) {
+                int nb_active = 0;
+
+                for (temp_sw = hw->sw_head.lh_first; temp_sw;
+                     temp_sw = temp_sw->entries.le_next) {
+                    nb_active += temp_sw->active != 0;
+                }
+
+                hw->pending_disable = nb_active == 1;
+            }
+        }
+
+        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+            sc->sw.active = hw->enabled;
+            if (hw->enabled) {
+                audio_capture_maybe_changed (sc->cap, 1);
+            }
+        }
+        sw->active = on;
+    }
+}
+
+void AUD_set_active_in (SWVoiceIn *sw, int on)
+{
+    HWVoiceIn *hw;
+
+    if (!sw) {
+        return;
+    }
+
+    hw = sw->hw;
+    if (sw->active != on) {
+        AudioState *s = sw->s;
+        SWVoiceIn *temp_sw;
+
+        if (on) {
+            if (!hw->enabled) {
+                hw->enabled = 1;
+                if (s->vm_running) {
+                    if (hw->pcm_ops->enable_in) {
+                        hw->pcm_ops->enable_in(hw, true);
+                    }
+                    audio_reset_timer (s);
+                }
+            }
+            sw->total_hw_samples_acquired = hw->total_samples_captured;
+        } else {
+            if (hw->enabled) {
+                int nb_active = 0;
+
+                for (temp_sw = hw->sw_head.lh_first; temp_sw;
+                     temp_sw = temp_sw->entries.le_next) {
+                    nb_active += temp_sw->active != 0;
+                }
+
+                if (nb_active == 1) {
+                    hw->enabled = 0;
+                    if (hw->pcm_ops->enable_in) {
+                        hw->pcm_ops->enable_in(hw, false);
+                    }
+                }
+            }
+        }
+        sw->active = on;
+    }
+}
+
+static size_t audio_get_avail (SWVoiceIn *sw)
+{
+    size_t live;
+
+    if (!sw) {
+        return 0;
+    }
+
+    live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+    if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
+        dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
+              sw->hw->conv_buf->size);
+        return 0;
+    }
+
+    ldebug (
+        "%s: get_avail live %zu ret %" PRId64 "\n",
+        SW_NAME (sw),
+        live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
+        );
+
+    return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
+}
+
+static size_t audio_get_free(SWVoiceOut *sw)
+{
+    size_t live, dead;
+
+    if (!sw) {
+        return 0;
+    }
+
+    live = sw->total_hw_samples_mixed;
+
+    if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
+        dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
+              sw->hw->mix_buf->size);
+        return 0;
+    }
+
+    dead = sw->hw->mix_buf->size - live;
+
+#ifdef DEBUG_OUT
+    dolog ("%s: get_free live %zu dead %zu ret %" PRId64 "\n",
+           SW_NAME (sw),
+           live, dead, (((int64_t) dead << 32) / sw->ratio) *
+           sw->info.bytes_per_frame);
+#endif
+
+    return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
+}
+
+static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
+                                        size_t samples)
+{
+    size_t n;
+
+    if (hw->enabled) {
+        SWVoiceCap *sc;
+
+        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+            SWVoiceOut *sw = &sc->sw;
+            int rpos2 = rpos;
+
+            n = samples;
+            while (n) {
+                size_t till_end_of_hw = hw->mix_buf->size - rpos2;
+                size_t to_write = MIN(till_end_of_hw, n);
+                size_t bytes = to_write * hw->info.bytes_per_frame;
+                size_t written;
+
+                sw->buf = hw->mix_buf->samples + rpos2;
+                written = audio_pcm_sw_write (sw, NULL, bytes);
+                if (written - bytes) {
+                    dolog("Could not mix %zu bytes into a capture "
+                          "buffer, mixed %zu\n",
+                          bytes, written);
+                    break;
+                }
+                n -= to_write;
+                rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
+            }
+        }
+    }
+
+    n = MIN(samples, hw->mix_buf->size - rpos);
+    mixeng_clear(hw->mix_buf->samples + rpos, n);
+    mixeng_clear(hw->mix_buf->samples, samples - n);
+}
+
+static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
+{
+    size_t clipped = 0;
+
+    while (live) {
+        size_t size = live * hw->info.bytes_per_frame;
+        size_t decr, proc;
+        void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
+
+        if (size == 0) {
+            break;
+        }
+
+        decr = MIN(size / hw->info.bytes_per_frame, live);
+        if (buf) {
+            audio_pcm_hw_clip_out(hw, buf, decr);
+        }
+        proc = hw->pcm_ops->put_buffer_out(hw, buf,
+                                           decr * hw->info.bytes_per_frame) /
+            hw->info.bytes_per_frame;
+
+        live -= proc;
+        clipped += proc;
+        hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
+
+        if (proc == 0 || proc < decr) {
+            break;
+        }
+    }
+
+    if (hw->pcm_ops->run_buffer_out) {
+        hw->pcm_ops->run_buffer_out(hw);
+    }
+
+    return clipped;
+}
+
+static void audio_run_out (AudioState *s)
+{
+    HWVoiceOut *hw = NULL;
+    SWVoiceOut *sw;
+
+    if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+        while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+            /* there is exactly 1 sw for each hw with no mixeng */
+            sw = hw->sw_head.lh_first;
+
+            if (hw->pending_disable) {
+                hw->enabled = 0;
+                hw->pending_disable = 0;
+                if (hw->pcm_ops->enable_out) {
+                    hw->pcm_ops->enable_out(hw, false);
+                }
+            }
+
+            if (sw->active) {
+                sw->callback.fn(sw->callback.opaque, INT_MAX);
+            }
+        }
+        return;
+    }
+
+    while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+        size_t played, live, prev_rpos, free;
+        int nb_live;
+
+        live = audio_pcm_hw_get_live_out (hw, &nb_live);
+        if (!nb_live) {
+            live = 0;
+        }
+
+        if (audio_bug(__func__, live > hw->mix_buf->size)) {
+            dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
+            continue;
+        }
+
+        if (hw->pending_disable && !nb_live) {
+            SWVoiceCap *sc;
+#ifdef DEBUG_OUT
+            dolog ("Disabling voice\n");
+#endif
+            hw->enabled = 0;
+            hw->pending_disable = 0;
+            if (hw->pcm_ops->enable_out) {
+                hw->pcm_ops->enable_out(hw, false);
+            }
+            for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+                sc->sw.active = 0;
+                audio_recalc_and_notify_capture (sc->cap);
+            }
+            continue;
+        }
+
+        if (!live) {
+            for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+                if (sw->active) {
+                    free = audio_get_free (sw);
+                    if (free > 0) {
+                        sw->callback.fn (sw->callback.opaque, free);
+                    }
+                }
+            }
+            if (hw->pcm_ops->run_buffer_out) {
+                hw->pcm_ops->run_buffer_out(hw);
+            }
+            continue;
+        }
+
+        prev_rpos = hw->mix_buf->pos;
+        played = audio_pcm_hw_run_out(hw, live);
+        replay_audio_out(&played);
+        if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
+            dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
+                  hw->mix_buf->pos, hw->mix_buf->size, played);
+            hw->mix_buf->pos = 0;
+        }
+
+#ifdef DEBUG_OUT
+        dolog("played=%zu\n", played);
+#endif
+
+        if (played) {
+            hw->ts_helper += played;
+            audio_capture_mix_and_clear (hw, prev_rpos, played);
+        }
+
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            if (!sw->active && sw->empty) {
+                continue;
+            }
+
+            if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
+                dolog("played=%zu sw->total_hw_samples_mixed=%zu\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) {
+                free = audio_get_free (sw);
+                if (free > 0) {
+                    sw->callback.fn (sw->callback.opaque, free);
+                }
+            }
+        }
+    }
+}
+
+static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
+{
+    size_t conv = 0;
+    STSampleBuffer *conv_buf = hw->conv_buf;
+
+    if (hw->pcm_ops->run_buffer_in) {
+        hw->pcm_ops->run_buffer_in(hw);
+    }
+
+    while (samples) {
+        size_t proc;
+        size_t size = samples * hw->info.bytes_per_frame;
+        void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
+
+        assert(size % hw->info.bytes_per_frame == 0);
+        if (size == 0) {
+            break;
+        }
+
+        proc = MIN(size / hw->info.bytes_per_frame,
+                   conv_buf->size - conv_buf->pos);
+
+        hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
+        conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
+
+        samples -= proc;
+        conv += proc;
+        hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
+    }
+
+    return conv;
+}
+
+static void audio_run_in (AudioState *s)
+{
+    HWVoiceIn *hw = NULL;
+
+    if (!audio_get_pdo_in(s->dev)->mixing_engine) {
+        while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
+            /* there is exactly 1 sw for each hw with no mixeng */
+            SWVoiceIn *sw = hw->sw_head.lh_first;
+            if (sw->active) {
+                sw->callback.fn(sw->callback.opaque, INT_MAX);
+            }
+        }
+        return;
+    }
+
+    while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
+        SWVoiceIn *sw;
+        size_t captured = 0, min;
+
+        if (replay_mode != REPLAY_MODE_PLAY) {
+            captured = audio_pcm_hw_run_in(
+                hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
+        }
+        replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
+                        hw->conv_buf->size);
+
+        min = audio_pcm_hw_find_min_in (hw);
+        hw->total_samples_captured += captured - min;
+        hw->ts_helper += captured;
+
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            sw->total_hw_samples_acquired -= min;
+
+            if (sw->active) {
+                size_t avail;
+
+                avail = audio_get_avail (sw);
+                if (avail > 0) {
+                    sw->callback.fn (sw->callback.opaque, avail);
+                }
+            }
+        }
+    }
+}
+
+static void audio_run_capture (AudioState *s)
+{
+    CaptureVoiceOut *cap;
+
+    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+        size_t live, rpos, captured;
+        HWVoiceOut *hw = &cap->hw;
+        SWVoiceOut *sw;
+
+        captured = live = audio_pcm_hw_get_live_out (hw, NULL);
+        rpos = hw->mix_buf->pos;
+        while (live) {
+            size_t left = hw->mix_buf->size - rpos;
+            size_t to_capture = MIN(live, left);
+            struct st_sample *src;
+            struct capture_callback *cb;
+
+            src = hw->mix_buf->samples + rpos;
+            hw->clip (cap->buf, src, to_capture);
+            mixeng_clear (src, to_capture);
+
+            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+                cb->ops.capture (cb->opaque, cap->buf,
+                                 to_capture * hw->info.bytes_per_frame);
+            }
+            rpos = (rpos + to_capture) % hw->mix_buf->size;
+            live -= to_capture;
+        }
+        hw->mix_buf->pos = rpos;
+
+        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+            if (!sw->active && sw->empty) {
+                continue;
+            }
+
+            if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
+                dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
+                      captured, sw->total_hw_samples_mixed);
+                captured = sw->total_hw_samples_mixed;
+            }
+
+            sw->total_hw_samples_mixed -= captured;
+            sw->empty = sw->total_hw_samples_mixed == 0;
+        }
+    }
+}
+
+void audio_run(AudioState *s, const char *msg)
+{
+    audio_run_out(s);
+    audio_run_in(s);
+    audio_run_capture(s);
+
+#ifdef DEBUG_POLL
+    {
+        static double prevtime;
+        double currtime;
+        struct timeval tv;
+
+        if (gettimeofday (&tv, NULL)) {
+            perror ("audio_run: gettimeofday");
+            return;
+        }
+
+        currtime = tv.tv_sec + tv.tv_usec * 1e-6;
+        dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
+        prevtime = currtime;
+    }
+#endif
+}
+
+void audio_generic_run_buffer_in(HWVoiceIn *hw)
+{
+    if (unlikely(!hw->buf_emul)) {
+        hw->size_emul = hw->samples * hw->info.bytes_per_frame;
+        hw->buf_emul = g_malloc(hw->size_emul);
+        hw->pos_emul = hw->pending_emul = 0;
+    }
+
+    while (hw->pending_emul < hw->size_emul) {
+        size_t read_len = MIN(hw->size_emul - hw->pos_emul,
+                              hw->size_emul - hw->pending_emul);
+        size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
+                                        read_len);
+        hw->pending_emul += read;
+        hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
+        if (read < read_len) {
+            break;
+        }
+    }
+}
+
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+    ssize_t start = (ssize_t)hw->pos_emul - hw->pending_emul;
+
+    if (start < 0) {
+        start += hw->size_emul;
+    }
+    assert(start >= 0 && start < hw->size_emul);
+
+    *size = MIN(*size, hw->pending_emul);
+    *size = MIN(*size, hw->size_emul - start);
+    return hw->buf_emul + start;
+}
+
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+    assert(size <= hw->pending_emul);
+    hw->pending_emul -= size;
+}
+
+void audio_generic_run_buffer_out(HWVoiceOut *hw)
+{
+    while (hw->pending_emul) {
+        size_t write_len, written;
+        ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+
+        if (start < 0) {
+            start += hw->size_emul;
+        }
+        assert(start >= 0 && start < hw->size_emul);
+
+        write_len = MIN(hw->pending_emul, hw->size_emul - start);
+
+        written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
+        hw->pending_emul -= written;
+
+        if (written < write_len) {
+            break;
+        }
+    }
+}
+
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+    if (unlikely(!hw->buf_emul)) {
+        hw->size_emul = hw->samples * hw->info.bytes_per_frame;
+        hw->buf_emul = g_malloc(hw->size_emul);
+        hw->pos_emul = hw->pending_emul = 0;
+    }
+
+    *size = MIN(hw->size_emul - hw->pending_emul,
+                hw->size_emul - hw->pos_emul);
+    return hw->buf_emul + hw->pos_emul;
+}
+
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+    assert(buf == hw->buf_emul + hw->pos_emul &&
+           size + hw->pending_emul <= hw->size_emul);
+
+    hw->pending_emul += size;
+    hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
+
+    return size;
+}
+
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
+{
+    size_t total = 0;
+
+    while (total < size) {
+        size_t dst_size = size - total;
+        size_t copy_size, proc;
+        void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
+
+        if (dst_size == 0) {
+            break;
+        }
+
+        copy_size = MIN(size - total, dst_size);
+        if (dst) {
+            memcpy(dst, (char *)buf + total, copy_size);
+        }
+        proc = hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
+        total += proc;
+
+        if (proc == 0 || proc < copy_size) {
+            break;
+        }
+    }
+
+    if (hw->pcm_ops->run_buffer_out) {
+        hw->pcm_ops->run_buffer_out(hw);
+    }
+
+    return total;
+}
+
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
+{
+    size_t total = 0;
+
+    if (hw->pcm_ops->run_buffer_in) {
+        hw->pcm_ops->run_buffer_in(hw);
+    }
+
+    while (total < size) {
+        size_t src_size = size - total;
+        void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
+
+        if (src_size == 0) {
+            break;
+        }
+
+        memcpy((char *)buf + total, src, src_size);
+        hw->pcm_ops->put_buffer_in(hw, src, src_size);
+        total += src_size;
+    }
+
+    return total;
+}
+
+static int audio_driver_init(AudioState *s, struct audio_driver *drv,
+                             bool msg, Audiodev *dev)
+{
+    s->drv_opaque = drv->init(dev);
+
+    if (s->drv_opaque) {
+        if (!drv->pcm_ops->get_buffer_in) {
+            drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
+            drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
+        }
+        if (!drv->pcm_ops->get_buffer_out) {
+            drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
+            drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
+        }
+
+        audio_init_nb_voices_out(s, drv);
+        audio_init_nb_voices_in(s, drv);
+        s->drv = drv;
+        return 0;
+    } else {
+        if (msg) {
+            dolog("Could not init `%s' audio driver\n", drv->name);
+        }
+        return -1;
+    }
+}
+
+static void audio_vm_change_state_handler (void *opaque, bool running,
+                                           RunState state)
+{
+    AudioState *s = opaque;
+    HWVoiceOut *hwo = NULL;
+    HWVoiceIn *hwi = NULL;
+
+    s->vm_running = running;
+    while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
+        if (hwo->pcm_ops->enable_out) {
+            hwo->pcm_ops->enable_out(hwo, running);
+        }
+    }
+
+    while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
+        if (hwi->pcm_ops->enable_in) {
+            hwi->pcm_ops->enable_in(hwi, running);
+        }
+    }
+    audio_reset_timer (s);
+}
+
+static void free_audio_state(AudioState *s)
+{
+    HWVoiceOut *hwo, *hwon;
+    HWVoiceIn *hwi, *hwin;
+
+    QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
+        SWVoiceCap *sc;
+
+        if (hwo->enabled && hwo->pcm_ops->enable_out) {
+            hwo->pcm_ops->enable_out(hwo, false);
+        }
+        hwo->pcm_ops->fini_out (hwo);
+
+        for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+            CaptureVoiceOut *cap = sc->cap;
+            struct capture_callback *cb;
+
+            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+                cb->ops.destroy (cb->opaque);
+            }
+        }
+        QLIST_REMOVE(hwo, entries);
+    }
+
+    QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
+        if (hwi->enabled && hwi->pcm_ops->enable_in) {
+            hwi->pcm_ops->enable_in(hwi, false);
+        }
+        hwi->pcm_ops->fini_in (hwi);
+        QLIST_REMOVE(hwi, entries);
+    }
+
+    if (s->drv) {
+        s->drv->fini (s->drv_opaque);
+        s->drv = NULL;
+    }
+
+    if (s->dev) {
+        qapi_free_Audiodev(s->dev);
+        s->dev = NULL;
+    }
+
+    if (s->ts) {
+        timer_free(s->ts);
+        s->ts = NULL;
+    }
+
+    g_free(s);
+}
+
+void audio_cleanup(void)
+{
+    while (!QTAILQ_EMPTY(&audio_states)) {
+        AudioState *s = QTAILQ_FIRST(&audio_states);
+        QTAILQ_REMOVE(&audio_states, s, list);
+        free_audio_state(s);
+    }
+}
+
+static bool vmstate_audio_needed(void *opaque)
+{
+    /*
+     * Never needed, this vmstate only exists in case
+     * an old qemu sends it to us.
+     */
+    return false;
+}
+
+static const VMStateDescription vmstate_audio = {
+    .name = "audio",
+    .version_id = 1,
+    .minimum_version_id = 1,
+    .needed = vmstate_audio_needed,
+    .fields = (VMStateField[]) {
+        VMSTATE_END_OF_LIST()
+    }
+};
+
+static void audio_validate_opts(Audiodev *dev, Error **errp);
+
+static AudiodevListEntry *audiodev_find(
+    AudiodevListHead *head, const char *drvname)
+{
+    AudiodevListEntry *e;
+    QSIMPLEQ_FOREACH(e, head, next) {
+        if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
+            return e;
+        }
+    }
+
+    return NULL;
+}
+
+/*
+ * if we have dev, this function was called because of an -audiodev argument =>
+ *   initialize a new state with it
+ * if dev == NULL => legacy implicit initialization, return the already created
+ *   state or create a new one
+ */
+static AudioState *audio_init(Audiodev *dev, const char *name)
+{
+    static bool atexit_registered;
+    size_t i;
+    int done = 0;
+    const char *drvname = NULL;
+    VMChangeStateEntry *e;
+    AudioState *s;
+    struct audio_driver *driver;
+    /* silence gcc warning about uninitialized variable */
+    AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
+
+    if (using_spice) {
+        /*
+         * When using spice allow the spice audio driver being picked
+         * as default.
+         *
+         * Temporary hack.  Using audio devices without explicit
+         * audiodev= property is already deprecated.  Same goes for
+         * the -soundhw switch.  Once this support gets finally
+         * removed we can also drop the concept of a default audio
+         * backend and this can go away.
+         */
+        driver = audio_driver_lookup("spice");
+        if (driver) {
+            driver->can_be_default = 1;
+        }
+    }
+
+    if (dev) {
+        /* -audiodev option */
+        legacy_config = false;
+        drvname = AudiodevDriver_str(dev->driver);
+    } else if (!QTAILQ_EMPTY(&audio_states)) {
+        if (!legacy_config) {
+            dolog("Device %s: audiodev default parameter is deprecated, please "
+                  "specify audiodev=%s\n", name,
+                  QTAILQ_FIRST(&audio_states)->dev->id);
+        }
+        return QTAILQ_FIRST(&audio_states);
+    } else {
+        /* legacy implicit initialization */
+        head = audio_handle_legacy_opts();
+        /*
+         * In case of legacy initialization, all Audiodevs in the list will have
+         * the same configuration (except the driver), so it doesn't matter which
+         * one we chose.  We need an Audiodev to set up AudioState before we can
+         * init a driver.  Also note that dev at this point is still in the
+         * list.
+         */
+        dev = QSIMPLEQ_FIRST(&head)->dev;
+        audio_validate_opts(dev, &error_abort);
+    }
+
+    s = g_malloc0(sizeof(AudioState));
+    s->dev = dev;
+
+    QLIST_INIT (&s->hw_head_out);
+    QLIST_INIT (&s->hw_head_in);
+    QLIST_INIT (&s->cap_head);
+    if (!atexit_registered) {
+        atexit(audio_cleanup);
+        atexit_registered = true;
+    }
+    QTAILQ_INSERT_TAIL(&audio_states, s, list);
+
+    s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
+
+    s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
+    s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
+
+    if (s->nb_hw_voices_out <= 0) {
+        dolog ("Bogus number of playback voices %d, setting to 1\n",
+               s->nb_hw_voices_out);
+        s->nb_hw_voices_out = 1;
+    }
+
+    if (s->nb_hw_voices_in <= 0) {
+        dolog ("Bogus number of capture voices %d, setting to 0\n",
+               s->nb_hw_voices_in);
+        s->nb_hw_voices_in = 0;
+    }
+
+    if (drvname) {
+        driver = audio_driver_lookup(drvname);
+        if (driver) {
+            done = !audio_driver_init(s, driver, true, dev);
+        } else {
+            dolog ("Unknown audio driver `%s'\n", drvname);
+        }
+    } else {
+        for (i = 0; audio_prio_list[i]; i++) {
+            AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
+            driver = audio_driver_lookup(audio_prio_list[i]);
+
+            if (e && driver) {
+                s->dev = dev = e->dev;
+                audio_validate_opts(dev, &error_abort);
+                done = !audio_driver_init(s, driver, false, dev);
+                if (done) {
+                    e->dev = NULL;
+                    break;
+                }
+            }
+        }
+    }
+    audio_free_audiodev_list(&head);
+
+    if (!done) {
+        driver = audio_driver_lookup("none");
+        done = !audio_driver_init(s, driver, false, dev);
+        assert(done);
+        dolog("warning: Using timer based audio emulation\n");
+    }
+
+    if (dev->timer_period <= 0) {
+        s->period_ticks = 1;
+    } else {
+        s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
+    }
+
+    e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
+    if (!e) {
+        dolog ("warning: Could not register change state handler\n"
+               "(Audio can continue looping even after stopping the VM)\n");
+    }
+
+    QLIST_INIT (&s->card_head);
+    vmstate_register (NULL, 0, &vmstate_audio, s);
+    return s;
+}
+
+void audio_free_audiodev_list(AudiodevListHead *head)
+{
+    AudiodevListEntry *e;
+    while ((e = QSIMPLEQ_FIRST(head))) {
+        QSIMPLEQ_REMOVE_HEAD(head, next);
+        qapi_free_Audiodev(e->dev);
+        g_free(e);
+    }
+}
+
+void AUD_register_card (const char *name, QEMUSoundCard *card)
+{
+    if (!card->state) {
+        card->state = audio_init(NULL, name);
+    }
+
+    card->name = g_strdup (name);
+    memset (&card->entries, 0, sizeof (card->entries));
+    QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
+}
+
+void AUD_remove_card (QEMUSoundCard *card)
+{
+    QLIST_REMOVE (card, entries);
+    g_free (card->name);
+}
+
+
+CaptureVoiceOut *AUD_add_capture(
+    AudioState *s,
+    struct audsettings *as,
+    struct audio_capture_ops *ops,
+    void *cb_opaque
+    )
+{
+    CaptureVoiceOut *cap;
+    struct capture_callback *cb;
+
+    if (!s) {
+        if (!legacy_config) {
+            dolog("Capturing without setting an audiodev is deprecated\n");
+        }
+        s = audio_init(NULL, NULL);
+    }
+
+    if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+        dolog("Can't capture with mixeng disabled\n");
+        return NULL;
+    }
+
+    if (audio_validate_settings (as)) {
+        dolog ("Invalid settings were passed when trying to add capture\n");
+        audio_print_settings (as);
+        return NULL;
+    }
+
+    cb = g_malloc0(sizeof(*cb));
+    cb->ops = *ops;
+    cb->opaque = cb_opaque;
+
+    cap = audio_pcm_capture_find_specific(s, as);
+    if (cap) {
+        QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+        return cap;
+    } else {
+        HWVoiceOut *hw;
+        CaptureVoiceOut *cap;
+
+        cap = g_malloc0(sizeof(*cap));
+
+        hw = &cap->hw;
+        hw->s = s;
+        QLIST_INIT (&hw->sw_head);
+        QLIST_INIT (&cap->cb_head);
+
+        /* XXX find a more elegant way */
+        hw->samples = 4096 * 4;
+        audio_pcm_hw_alloc_resources_out(hw);
+
+        audio_pcm_init_info (&hw->info, as);
+
+        cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
+
+        if (hw->info.is_float) {
+            hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
+        } else {
+            hw->clip = mixeng_clip
+                [hw->info.nchannels == 2]
+                [hw->info.is_signed]
+                [hw->info.swap_endianness]
+                [audio_bits_to_index(hw->info.bits)];
+        }
+
+        QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
+        QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+
+        QLIST_FOREACH(hw, &s->hw_head_out, entries) {
+            audio_attach_capture (hw);
+        }
+        return cap;
+    }
+}
+
+void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
+{
+    struct capture_callback *cb;
+
+    for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+        if (cb->opaque == cb_opaque) {
+            cb->ops.destroy (cb_opaque);
+            QLIST_REMOVE (cb, entries);
+            g_free (cb);
+
+            if (!cap->cb_head.lh_first) {
+                SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
+
+                while (sw) {
+                    SWVoiceCap *sc = (SWVoiceCap *) sw;
+#ifdef DEBUG_CAPTURE
+                    dolog ("freeing %s\n", sw->name);
+#endif
+
+                    sw1 = sw->entries.le_next;
+                    if (sw->rate) {
+                        st_rate_stop (sw->rate);
+                        sw->rate = NULL;
+                    }
+                    QLIST_REMOVE (sw, entries);
+                    QLIST_REMOVE (sc, entries);
+                    g_free (sc);
+                    sw = sw1;
+                }
+                QLIST_REMOVE (cap, entries);
+                g_free (cap->hw.mix_buf);
+                g_free (cap->buf);
+                g_free (cap);
+            }
+            return;
+        }
+    }
+}
+
+void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+    audio_set_volume_out(sw, &vol);
+}
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
+{
+    if (sw) {
+        HWVoiceOut *hw = sw->hw;
+
+        sw->vol.mute = vol->mute;
+        sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+        sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
+            255;
+
+        if (hw->pcm_ops->volume_out) {
+            hw->pcm_ops->volume_out(hw, vol);
+        }
+    }
+}
+
+void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+    audio_set_volume_in(sw, &vol);
+}
+
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
+{
+    if (sw) {
+        HWVoiceIn *hw = sw->hw;
+
+        sw->vol.mute = vol->mute;
+        sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+        sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
+            255;
+
+        if (hw->pcm_ops->volume_in) {
+            hw->pcm_ops->volume_in(hw, vol);
+        }
+    }
+}
+
+void audio_create_pdos(Audiodev *dev)
+{
+    switch (dev->driver) {
+#define CASE(DRIVER, driver, pdo_name)                              \
+    case AUDIODEV_DRIVER_##DRIVER:                                  \
+        if (!dev->u.driver.has_in) {                                \
+            dev->u.driver.in = g_malloc0(                           \
+                sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
+            dev->u.driver.has_in = true;                            \
+        }                                                           \
+        if (!dev->u.driver.has_out) {                               \
+            dev->u.driver.out = g_malloc0(                          \
+                sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
+            dev->u.driver.has_out = true;                           \
+        }                                                           \
+        break
+
+        CASE(NONE, none, );
+        CASE(ALSA, alsa, Alsa);
+        CASE(COREAUDIO, coreaudio, Coreaudio);
+        CASE(DSOUND, dsound, );
+        CASE(JACK, jack, Jack);
+        CASE(OSS, oss, Oss);
+        CASE(PA, pa, Pa);
+        CASE(SDL, sdl, Sdl);
+        CASE(SPICE, spice, );
+        CASE(WAV, wav, );
+
+    case AUDIODEV_DRIVER__MAX:
+        abort();
+    };
+}
+
+static void audio_validate_per_direction_opts(
+    AudiodevPerDirectionOptions *pdo, Error **errp)
+{
+    if (!pdo->has_mixing_engine) {
+        pdo->has_mixing_engine = true;
+        pdo->mixing_engine = true;
+    }
+    if (!pdo->has_fixed_settings) {
+        pdo->has_fixed_settings = true;
+        pdo->fixed_settings = pdo->mixing_engine;
+    }
+    if (!pdo->fixed_settings &&
+        (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
+        error_setg(errp,
+                   "You can't use frequency, channels or format with fixed-settings=off");
+        return;
+    }
+    if (!pdo->mixing_engine && pdo->fixed_settings) {
+        error_setg(errp, "You can't use fixed-settings without mixeng");
+        return;
+    }
+
+    if (!pdo->has_frequency) {
+        pdo->has_frequency = true;
+        pdo->frequency = 44100;
+    }
+    if (!pdo->has_channels) {
+        pdo->has_channels = true;
+        pdo->channels = 2;
+    }
+    if (!pdo->has_voices) {
+        pdo->has_voices = true;
+        pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
+    }
+    if (!pdo->has_format) {
+        pdo->has_format = true;
+        pdo->format = AUDIO_FORMAT_S16;
+    }
+}
+
+static void audio_validate_opts(Audiodev *dev, Error **errp)
+{
+    Error *err = NULL;
+
+    audio_create_pdos(dev);
+
+    audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
+    if (err) {
+        error_propagate(errp, err);
+        return;
+    }
+
+    audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
+    if (err) {
+        error_propagate(errp, err);
+        return;
+    }
+
+    if (!dev->has_timer_period) {
+        dev->has_timer_period = true;
+        dev->timer_period = 10000; /* 100Hz -> 10ms */
+    }
+}
+
+void audio_parse_option(const char *opt)
+{
+    AudiodevListEntry *e;
+    Audiodev *dev = NULL;
+
+    Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
+    visit_type_Audiodev(v, NULL, &dev, &error_fatal);
+    visit_free(v);
+
+    audio_validate_opts(dev, &error_fatal);
+
+    e = g_malloc0(sizeof(AudiodevListEntry));
+    e->dev = dev;
+    QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
+}
+
+void audio_init_audiodevs(void)
+{
+    AudiodevListEntry *e;
+
+    QSIMPLEQ_FOREACH(e, &audiodevs, next) {
+        audio_init(e->dev, NULL);
+    }
+}
+
+audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
+{
+    return (audsettings) {
+        .freq = pdo->frequency,
+        .nchannels = pdo->channels,
+        .fmt = pdo->format,
+        .endianness = AUDIO_HOST_ENDIANNESS,
+    };
+}
+
+int audioformat_bytes_per_sample(AudioFormat fmt)
+{
+    switch (fmt) {
+    case AUDIO_FORMAT_U8:
+    case AUDIO_FORMAT_S8:
+        return 1;
+
+    case AUDIO_FORMAT_U16:
+    case AUDIO_FORMAT_S16:
+        return 2;
+
+    case AUDIO_FORMAT_U32:
+    case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_F32:
+        return 4;
+
+    case AUDIO_FORMAT__MAX:
+        ;
+    }
+    abort();
+}
+
+
+/* frames = freq * usec / 1e6 */
+int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
+                        audsettings *as, int def_usecs)
+{
+    uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
+    return (as->freq * usecs + 500000) / 1000000;
+}
+
+/* samples = channels * frames = channels * freq * usec / 1e6 */
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+                         audsettings *as, int def_usecs)
+{
+    return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
+}
+
+/*
+ * bytes = bytes_per_sample * samples =
+ *     bytes_per_sample * channels * freq * usec / 1e6
+ */
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+                       audsettings *as, int def_usecs)
+{
+    return audio_buffer_samples(pdo, as, def_usecs) *
+        audioformat_bytes_per_sample(as->fmt);
+}
+
+AudioState *audio_state_by_name(const char *name)
+{
+    AudioState *s;
+    QTAILQ_FOREACH(s, &audio_states, list) {
+        assert(s->dev);
+        if (strcmp(name, s->dev->id) == 0) {
+            return s;
+        }
+    }
+    return NULL;
+}
+
+const char *audio_get_id(QEMUSoundCard *card)
+{
+    if (card->state) {
+        assert(card->state->dev);
+        return card->state->dev->id;
+    } else {
+        return "";
+    }
+}
+
+const char *audio_application_name(void)
+{
+    const char *vm_name;
+
+    vm_name = qemu_get_vm_name();
+    return vm_name ? vm_name : "qemu";
+}
+
+void audio_rate_start(RateCtl *rate)
+{
+    memset(rate, 0, sizeof(RateCtl));
+    rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+}
+
+size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
+                            size_t bytes_avail)
+{
+    int64_t now;
+    int64_t ticks;
+    int64_t bytes;
+    int64_t samples;
+    size_t ret;
+
+    now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+    ticks = now - rate->start_ticks;
+    bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
+    samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
+    if (samples < 0 || samples > 65536) {
+        AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
+        audio_rate_start(rate);
+        samples = 0;
+    }
+
+    ret = MIN(samples * info->bytes_per_frame, bytes_avail);
+    rate->bytes_sent += ret;
+    return ret;
+}