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audio.c
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/*
* 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 "sysemu/sysemu.h"
#include "qemu/cutils.h"
#include "qemu/module.h"
#include "sysemu/replay.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 AudioState glob_audio_state;
const struct mixeng_volume nominal_volume = {
.mute = 0,
#ifdef FLOAT_MIXENG
.r = 1.0,
.l = 1.0,
#else
.r = 1ULL << 32,
.l = 1ULL << 32,
#endif
};
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
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");
#if defined AUDIO_BREAKPOINT_ON_BUG
# if defined HOST_I386
# if defined __GNUC__
__asm__ ("int3");
# elif defined _MSC_VER
_asm _emit 0xcc;
# else
abort ();
# endif
# else
abort ();
# endif
#endif
}
return cond;
}
#endif
static inline int audio_bits_to_index (int bits)
{
switch (bits) {
case 8:
return 0;
case 16:
return 1;
case 32:
return 2;
default:
audio_bug ("bits_to_index", 1);
AUD_log (NULL, "invalid bits %d\n", bits);
return 0;
}
}
void *audio_calloc (const char *funcname, int nmemb, size_t size)
{
int cond;
size_t len;
len = nmemb * size;
cond = !nmemb || !size;
cond |= nmemb < 0;
cond |= len < size;
if (audio_bug ("audio_calloc", cond)) {
AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
funcname);
AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
return NULL;
}
return g_malloc0 (len);
}
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;
default:
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
}
AUD_log (NULL, " endianness=");
switch (as->endianness) {
case 0:
AUD_log (NULL, "little");
break;
case 1:
AUD_log (NULL, "big");
break;
default:
AUD_log (NULL, "invalid");
break;
}
AUD_log (NULL, "\n");
}
static int audio_validate_settings (struct audsettings *as)
{
int invalid;
invalid = as->nchannels != 1 && as->nchannels != 2;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
case AUDIO_FORMAT_U8:
case AUDIO_FORMAT_S16:
case AUDIO_FORMAT_U16:
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_U32:
break;
default:
invalid = 1;
break;
}
invalid |= as->freq <= 0;
return invalid ? -1 : 0;
}
static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
case AUDIO_FORMAT_U8:
break;
case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
break;
case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
break;
default:
abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->sign == sign
&& info->bits == bits
&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
sign = 1;
case AUDIO_FORMAT_U8:
break;
case AUDIO_FORMAT_S16:
sign = 1;
case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
case AUDIO_FORMAT_S32:
sign = 1;
case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
default:
abort();
}
info->freq = as->freq;
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
info->shift = (as->nchannels == 2) + shift;
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
if (!len) {
return;
}
if (info->sign) {
memset (buf, 0x00, len << info->shift);
}
else {
switch (info->bits) {
case 8:
memset (buf, 0x80, len << info->shift);
break;
case 16:
{
int i;
uint16_t *p = buf;
int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
case 32:
{
int i;
uint32_t *p = buf;
int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
default:
AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
info->bits);
break;
}
}
}
/*
* Capture
*/
static void noop_conv (struct st_sample *dst, const void *src, int samples)
{
(void) src;
(void) dst;
(void) samples;
}
static CaptureVoiceOut *audio_pcm_capture_find_specific (
struct audsettings *as
)
{
CaptureVoiceOut *cap;
AudioState *s = &glob_audio_state;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
if (audio_pcm_info_eq (&cap->hw.info, as)) {
return cap;
}
}
return NULL;
}
static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
{
struct capture_callback *cb;
#ifdef DEBUG_CAPTURE
dolog ("notification %d sent\n", cmd);
#endif
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.notify (cb->opaque, cmd);
}
}
static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
if (cap->hw.enabled != enabled) {
audcnotification_e cmd;
cap->hw.enabled = enabled;
cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
audio_notify_capture (cap, cmd);
}
}
static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
{
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
int enabled = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
enabled = 1;
break;
}
}
audio_capture_maybe_changed (cap, enabled);
}
static void audio_detach_capture (HWVoiceOut *hw)
{
SWVoiceCap *sc = hw->cap_head.lh_first;
while (sc) {
SWVoiceCap *sc1 = sc->entries.le_next;
SWVoiceOut *sw = &sc->sw;
CaptureVoiceOut *cap = sc->cap;
int was_active = sw->active;
if (sw->rate) {
st_rate_stop (sw->rate);
sw->rate = NULL;
}
QLIST_REMOVE (sw, entries);
QLIST_REMOVE (sc, entries);
g_free (sc);
if (was_active) {
/* We have removed soft voice from the capture:
this might have changed the overall status of the capture
since this might have been the only active voice */
audio_recalc_and_notify_capture (cap);
}
sc = sc1;
}
}
static int audio_attach_capture (HWVoiceOut *hw)
{
AudioState *s = &glob_audio_state;
CaptureVoiceOut *cap;
audio_detach_capture (hw);
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
SWVoiceCap *sc;
SWVoiceOut *sw;
HWVoiceOut *hw_cap = &cap->hw;
sc = 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 int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
int m = hw->total_samples_captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
m = audio_MIN (m, sw->total_hw_samples_acquired);
}
}
return m;
}
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
{
int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
if (audio_bug(__func__, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int live, int pending)
{
int left = hw->samples - pending;
int len = audio_MIN (left, live);
int clipped = 0;
while (len) {
struct st_sample *src = hw->mix_buf + hw->rpos;
uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
int samples_till_end_of_buf = hw->samples - hw->rpos;
int samples_to_clip = audio_MIN (len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
len -= samples_to_clip;
clipped += samples_to_clip;
}
return clipped;
}
/*
* Soft voice (capture)
*/
static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
{
HWVoiceIn *hw = sw->hw;
int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
int rpos;
if (audio_bug(__func__, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
rpos = hw->wpos - live;
if (rpos >= 0) {
return rpos;
}
else {
return hw->samples + rpos;
}
}
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
{
HWVoiceIn *hw = sw->hw;
int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug(__func__, live < 0 || live > hw->samples)) {
dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
samples = size >> sw->info.shift;
if (!live) {
return 0;
}
swlim = (live * sw->ratio) >> 32;
swlim = audio_MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf + rpos;
isamp = hw->wpos - rpos;
/* XXX: <= ? */
if (isamp <= 0) {
isamp = hw->samples - rpos;
}
if (!isamp) {
break;
}
osamp = swlim;
if (audio_bug(__func__, osamp < 0)) {
dolog ("osamp=%d\n", osamp);
return 0;
}
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->samples;
dst += osamp;
ret += osamp;
total += isamp;
}
if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
mixeng_volume (sw->buf, ret, &sw->vol);
}
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
return ret << sw->info.shift;
}
/*
* Hard voice (playback)
*/
static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
SWVoiceOut *sw;
int m = INT_MAX;
int nb_live = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active || !sw->empty) {
m = audio_MIN (m, sw->total_hw_samples_mixed);
nb_live += 1;
}
}
*nb_livep = nb_live;
return m;
}
static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
{
int smin;
int nb_live1;
smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
if (nb_live) {
*nb_live = nb_live1;
}
if (nb_live1) {
int live = smin;
if (audio_bug(__func__, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
return 0;
}
/*
* Soft voice (playback)
*/
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
{
int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
int ret = 0, pos = 0, total = 0;
if (!sw) {
return size;
}
hwsamples = sw->hw->samples;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live < 0 || live > hwsamples)) {
dolog ("live=%d hw->samples=%d\n", live, hwsamples);
return 0;
}
if (live == hwsamples) {
#ifdef DEBUG_OUT
dolog ("%s is full %d\n", sw->name, live);
#endif
return 0;
}
wpos = (sw->hw->rpos + live) % hwsamples;
samples = size >> sw->info.shift;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
swlim = audio_MIN (swlim, samples);
if (swlim) {
sw->conv (sw->buf, buf, swlim);
if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
mixeng_volume (sw->buf, swlim, &sw->vol);
}
}
while (swlim) {
dead = hwsamples - live;
left = hwsamples - wpos;
blck = audio_MIN (dead, left);
if (!blck) {
break;
}
isamp = swlim;
osamp = blck;
st_rate_flow_mix (
sw->rate,
sw->buf + pos,
sw->hw->mix_buf + wpos,
&isamp,
&osamp
);
ret += isamp;
swlim -= isamp;
pos += isamp;
live += osamp;
wpos = (wpos + osamp) % hwsamples;
total += osamp;
}
sw->total_hw_samples_mixed += total;
sw->empty = sw->total_hw_samples_mixed == 0;
#ifdef DEBUG_OUT
dolog (
"%s: write size %d ret %d total sw %d\n",
SW_NAME (sw),
size >> sw->info.shift,
ret,
sw->total_hw_samples_mixed
);
#endif
return ret << sw->info.shift;
}
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
{
dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
cap, info->bits, info->sign, info->freq, info->nchannels);
}
#endif
#define DAC
#include "audio_template.h"
#undef DAC
#include "audio_template.h"
/*
* Timer
*/
static bool audio_timer_running;
static uint64_t audio_timer_last;
static int audio_is_timer_needed (void)
{
HWVoiceIn *hwi = NULL;
HWVoiceOut *hwo = NULL;
while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
if (!hwo->poll_mode) return 1;
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
if (!hwi->poll_mode) return 1;
}
return 0;
}
static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed ()) {
timer_mod_anticipate_ns(s->ts,
qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
if (!audio_timer_running) {
audio_timer_running = true;
audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
if (audio_timer_running) {
audio_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 - audio_timer_last;
if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
audio_timer_last = now;
audio_run("timer");
audio_reset_timer(s);
}
/*
* Public API
*/
int AUD_write (SWVoiceOut *sw, void *buf, int size)
{
if (!sw) {
/* XXX: Consider options */
return size;
}
if (!sw->hw->enabled) {
dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
return sw->hw->pcm_ops->write(sw, buf, size);
}
int AUD_read (SWVoiceIn *sw, void *buf, int size)
{
if (!sw) {
/* XXX: Consider options */
return size;
}
if (!sw->hw->enabled) {
dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
return sw->hw->pcm_ops->read(sw, buf, size);
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
return sw->hw->samples << sw->hw->info.shift;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
{
HWVoiceOut *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioState *s = &glob_audio_state;
SWVoiceOut *temp_sw;
SWVoiceCap *sc;
if (on) {
hw->pending_disable = 0;
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
hw->pcm_ops->ctl_out(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
}
else {
if (hw->enabled) {
int nb_active = 0;
for (temp_sw = hw->sw_head.lh_first; temp_sw;
temp_sw = temp_sw->entries.le_next) {
nb_active += temp_sw->active != 0;
}
hw->pending_disable = nb_active == 1;
}
}
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = hw->enabled;
if (hw->enabled) {
audio_capture_maybe_changed (sc->cap, 1);
}
}
sw->active = on;
}
}
void AUD_set_active_in (SWVoiceIn *sw, int on)
{
HWVoiceIn *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioState *s = &glob_audio_state;
SWVoiceIn *temp_sw;
if (on) {
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
hw->pcm_ops->ctl_in(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
sw->total_hw_samples_acquired = hw->total_samples_captured;
}
else {
if (hw->enabled) {
int nb_active = 0;
for (temp_sw = hw->sw_head.lh_first; temp_sw;
temp_sw = temp_sw->entries.le_next) {
nb_active += temp_sw->active != 0;
}
if (nb_active == 1) {
hw->enabled = 0;
hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
}
}
}
sw->active = on;
}
}
static int audio_get_avail (SWVoiceIn *sw)
{
int live;
if (!sw) {
return 0;
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
return 0;
}
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
);
return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
}
static int audio_get_free (SWVoiceOut *sw)
{
int live, dead;
if (!sw) {
return 0;