qemu

audio.c (61008B)

---

 /*
  * 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/help_option.h"
 #include "sysemu/sysemu.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;
     Error *local_err = NULL;
     int rv;
 
     QLIST_FOREACH(d, &audio_drivers, next) {
         if (strcmp(name, d->name) == 0) {
             return d;
         }
     }
     rv = audio_module_load(name, &local_err);
     if (rv > 0) {
         QLIST_FOREACH(d, &audio_drivers, next) {
             if (strcmp(name, d->name) == 0) {
                 return d;
             }
         }
     } else if (rv < 0) {
         error_report_err(local_err);
     }
     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");
     }
 
     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 size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
 {
     size_t conv = 0;
     STSampleBuffer *conv_buf = hw->conv_buf;
 
     while (samples) {
         uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
         size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
 
         hw->conv(conv_buf->samples + conv_buf->pos, src, proc);
         conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
         samples -= proc;
         conv += proc;
     }
 
     return conv;
 }
 
 /*
  * Soft voice (capture)
  */
 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;
 
     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
     if (!live) {
         return 0;
     }
     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;
     }
 
     rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size);
 
     samples = size / sw->info.bytes_per_frame;
 
     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->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;
 }
 
 static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
 {
     return (hw->pcm_ops->buffer_get_free ? hw->pcm_ops->buffer_get_free(hw) :
             INT_MAX) / hw->info.bytes_per_frame;
 }
 
 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 (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, blck;
     size_t hw_free;
     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;
 
     dead = hwsamples - live;
     hw_free = audio_pcm_hw_get_free(sw->hw);
     hw_free = hw_free > live ? hw_free - live : 0;
     samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
     samples = MIN(samples, size / sw->info.bytes_per_frame);
     if (samples) {
         sw->conv(sw->buf, buf, samples);
 
         if (!sw->hw->pcm_ops->volume_out) {
             mixeng_volume(sw->buf, samples, &sw->vol);
         }
     }
 
     while (samples) {
         dead = hwsamples - live;
         left = hwsamples - wpos;
         blck = MIN (dead, left);
         if (!blck) {
             break;
         }
         isamp = samples;
         osamp = blck;
         st_rate_flow_mix (
             sw->rate,
             sw->buf + pos,
             sw->hw->mix_buf->samples + wpos,
             &isamp,
             &osamp
             );
         ret += isamp;
         samples -= 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;
     }
 }
 
 /**
  * audio_frontend_frames_in() - returns the number of frames the resampling
  * code generates from frames_in frames
  *
  * @sw: audio recording frontend
  * @frames_in: number of frames
  */
 static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in)
 {
     return (int64_t)frames_in * sw->ratio >> 32;
 }
 
 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 frontend frames %zu\n",
         SW_NAME (sw),
         live, audio_frontend_frames_in(sw, live)
         );
 
     return live;
 }
 
 /**
  * audio_frontend_frames_out() - returns the number of frames needed to
  * get frames_out frames after resampling
  *
  * @sw: audio playback frontend
  * @frames_out: number of frames
  */
 static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out)
 {
     return ((int64_t)frames_out << 32) / sw->ratio;
 }
 
 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 frontend frames %zu\n",
           SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
 #endif
 
     return dead;
 }
 
 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;
 
     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
         size_t played, live, prev_rpos;
         size_t hw_free = audio_pcm_hw_get_free(hw);
         int nb_live;
 
         if (!audio_get_pdo_out(s->dev)->mixing_engine) {
             /* 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,
                                 hw_free * sw->info.bytes_per_frame);
             }
 
             if (hw->pcm_ops->run_buffer_out) {
                 hw->pcm_ops->run_buffer_out(hw);
             }
 
             continue;
         }
 
         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
             if (sw->active) {
                 size_t sw_free = audio_get_free(sw);
                 size_t free;
 
                 if (hw_free > sw->total_hw_samples_mixed) {
                     free = audio_frontend_frames_out(sw,
                         MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
                 } else {
                     free = 0;
                 }
                 if (free > 0) {
                     sw->callback.fn(sw->callback.opaque,
                                     free * sw->info.bytes_per_frame);
                 }
             }
         }
 
         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) {
             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;
             }
         }
     }
 }
 
 static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
 {
     size_t conv = 0;
 
     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 = audio_pcm_hw_conv_in(hw, buf, size / hw->info.bytes_per_frame);
 
         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 sw_avail = audio_get_avail(sw);
                 size_t avail;
 
                 avail = audio_frontend_frames_in(sw, sw_avail);
                 if (avail > 0) {
                     sw->callback.fn(sw->callback.opaque,
                                     avail * sw->info.bytes_per_frame);
                 }
             }
         }
     }
 }
 
 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)
 {
     size_t start;
 
     start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
     assert(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;
 }
 
 size_t audio_generic_buffer_get_free(HWVoiceOut *hw)
 {
     if (hw->buf_emul) {
         return hw->size_emul - hw->pending_emul;
     } else {
         return hw->samples * hw->info.bytes_per_frame;
     }
 }
 
 void audio_generic_run_buffer_out(HWVoiceOut *hw)
 {
     while (hw->pending_emul) {
         size_t write_len, written, start;
 
         start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
         assert(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;
 
     if (hw->pcm_ops->buffer_get_free) {
         size_t free = hw->pcm_ops->buffer_get_free(hw);
 
         size = MIN(size, free);
     }
 
     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;
         }
     }
 
     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_new0(AudioState, 1);
     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;
     }
 
     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 < 1) {
         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);
         }
         if (!done) {
             free_audio_state(s);
             return NULL;
         }
     } 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");
     }
 
     QTAILQ_INSERT_TAIL(&audio_states, s, list);
     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);
 }
 
 static struct audio_pcm_ops capture_pcm_ops;
 
 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;
         hw->pcm_ops = &capture_pcm_ops;
         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(DBUS, dbus, );
         CASE(DSOUND, dsound, );
         CASE(JACK, jack, Jack);
         CASE(OSS, oss, Oss);
         CASE(PA, pa, Pa);
         CASE(SDL, sdl, Sdl);
         CASE(SNDIO, sndio, );
         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_help(void)
 {
     int i;
 
     printf("Available audio drivers:\n");
 
     for (i = 0; i < AUDIODEV_DRIVER__MAX; i++) {
         audio_driver *driver = audio_driver_lookup(AudiodevDriver_str(i));
         if (driver) {
             printf("%s\n", driver->name);
         }
     }
 }
 
 void audio_parse_option(const char *opt)
 {
     Audiodev *dev = NULL;
 
     if (is_help_option(opt)) {
         audio_help();
         exit(EXIT_SUCCESS);
     }
     Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
     visit_type_Audiodev(v, NULL, &dev, &error_fatal);
     visit_free(v);
 
     audio_define(dev);
 }
 
 void audio_define(Audiodev *dev)
 {
     AudiodevListEntry *e;
 
     audio_validate_opts(dev, &error_fatal);
 
     e = g_new0(AudiodevListEntry, 1);
     e->dev = dev;
     QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
 }
 
 bool audio_init_audiodevs(void)
 {
     AudiodevListEntry *e;
 
     QSIMPLEQ_FOREACH(e, &audiodevs, next) {
         if (!audio_init(e->dev, NULL)) {
             return false;
         }
     }
 
     return true;
 }
 
 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_peek_bytes(RateCtl *rate, struct audio_pcm_info *info)
 {
     int64_t now;
     int64_t ticks;
     int64_t bytes;
     int64_t frames;
 
     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
     ticks = now - rate->start_ticks;
     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
     frames = (bytes - rate->bytes_sent) / info->bytes_per_frame;
     if (frames < 0 || frames > 65536) {
         AUD_log(NULL, "Resetting rate control (%" PRId64 " frames)\n", frames);
         audio_rate_start(rate);
         frames = 0;
     }
 
     return frames * info->bytes_per_frame;
 }
 
 void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used)
 {
     rate->bytes_sent += bytes_used;
 }
 
 size_t audio_rate_get_bytes(RateCtl *rate, struct audio_pcm_info *info,
                             size_t bytes_avail)
 {
     size_t bytes;
 
     bytes = audio_rate_peek_bytes(rate, info);
     bytes = MIN(bytes, bytes_avail);
     audio_rate_add_bytes(rate, bytes);
 
     return bytes;
 }