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qemu-e2k/audio/audio.c
aliguori 376253ece4 monitor: Rework API (Jan Kiszka) 
Refactor the monitor API and prepare it for decoupled terminals:
term_print functions are renamed to monitor_* and all monitor services
gain a new parameter (mon) that will once refer to the monitor instance
the output is supposed to appear on. However, the argument remains
unused for now. All monitor command callbacks are also extended by a mon
parameter so that command handlers are able to pass an appropriate
reference to monitor output services.

For the case that monitor outputs so far happen without clearly
identifiable context, the global variable cur_mon is introduced that
shall once provide a pointer either to the current active monitor (while
processing commands) or to the default one. On the mid or long term,
those use case will be obsoleted so that this variable can be removed
again.

Due to the broad usage of the monitor interface, this patch mostly deals
with converting users of the monitor API. A few of them are already
extended to pass 'mon' from the command handler further down to internal
functions that invoke monitor_printf.

At this chance, monitor-related prototypes are moved from console.h to
a new monitor.h. The same is done for the readline API.

Signed-off-by: Jan Kiszka <jan.kiszka@siemens.com>
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>


git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@6711 c046a42c-6fe2-441c-8c8c-71466251a162
2009-03-05 23:01:23 +00:00

1964 lines
48 KiB
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 "hw/hw.h"
#include "audio.h"
#include "monitor.h"
#include "qemu-timer.h"
#include "sysemu.h"
#define AUDIO_CAP "audio"
#include "audio_int.h"
/* #define DEBUG_PLIVE */
/* #define DEBUG_LIVE */
/* #define DEBUG_OUT */
/* #define DEBUG_CAPTURE */
#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
static struct audio_driver *drvtab[] = {
AUDIO_DRIVERS
&no_audio_driver,
&wav_audio_driver
};
struct fixed_settings {
int enabled;
int nb_voices;
int greedy;
struct audsettings settings;
};
static struct {
struct fixed_settings fixed_out;
struct fixed_settings fixed_in;
union {
int hertz;
int64_t ticks;
} period;
int plive;
int log_to_monitor;
} conf = {
{ /* DAC fixed settings */
1, /* enabled */
1, /* nb_voices */
1, /* greedy */
{
44100, /* freq */
2, /* nchannels */
AUD_FMT_S16, /* fmt */
AUDIO_HOST_ENDIANNESS
}
},
{ /* ADC fixed settings */
1, /* enabled */
1, /* nb_voices */
1, /* greedy */
{
44100, /* freq */
2, /* nchannels */
AUD_FMT_S16, /* fmt */
AUDIO_HOST_ENDIANNESS
}
},
{ 250 }, /* period */
0, /* plive */
0 /* log_to_monitor */
};
static AudioState glob_audio_state;
struct mixeng_volume nominal_volume = {
0,
#ifdef FLOAT_MIXENG
1.0,
1.0
#else
1ULL << 32,
1ULL << 32
#endif
};
/* http://www.df.lth.se/~john_e/gems/gem002d.html */
/* http://www.multi-platforms.com/Tips/PopCount.htm */
uint32_t popcount (uint32_t u)
{
u = ((u&0x55555555) + ((u>>1)&0x55555555));
u = ((u&0x33333333) + ((u>>2)&0x33333333));
u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
u = ( u&0x0000ffff) + (u>>16);
return u;
}
inline uint32_t lsbindex (uint32_t u)
{
return popcount ((u&-u)-1);
}
#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, "Please send bug report to malc@pulsesoft.com\n");
AUD_log (NULL, "I am sorry\n");
}
AUD_log (NULL, "Context:\n");
#if defined AUDIO_BREAKPOINT_ON_BUG
# if defined HOST_I386
# if defined __GNUC__
__asm__ ("int3");
# elif defined _MSC_VER
_asm _emit 0xcc;
# else
abort ();
# endif
# else
abort ();
# endif
#endif
}
return cond;
}
#endif
static 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 qemu_mallocz (len);
}
static char *audio_alloc_prefix (const char *s)
{
const char qemu_prefix[] = "QEMU_";
size_t len, i;
char *r, *u;
if (!s) {
return NULL;
}
len = strlen (s);
r = qemu_malloc (len + sizeof (qemu_prefix));
u = r + sizeof (qemu_prefix) - 1;
pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
pstrcat (r, len + sizeof (qemu_prefix), s);
for (i = 0; i < len; ++i) {
u[i] = qemu_toupper(u[i]);
}
return r;
}
static const char *audio_audfmt_to_string (audfmt_e fmt)
{
switch (fmt) {
case AUD_FMT_U8:
return "U8";
case AUD_FMT_U16:
return "U16";
case AUD_FMT_S8:
return "S8";
case AUD_FMT_S16:
return "S16";
case AUD_FMT_U32:
return "U32";
case AUD_FMT_S32:
return "S32";
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
return AUD_FMT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
return AUD_FMT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
return AUD_FMT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
return AUD_FMT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
return AUD_FMT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
return AUD_FMT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
s, audio_audfmt_to_string (defval));
*defaultp = 1;
return defval;
}
}
static audfmt_e audio_get_conf_fmt (const char *envname,
audfmt_e defval,
int *defaultp)
{
const char *var = getenv (envname);
if (!var) {
*defaultp = 1;
return defval;
}
return audio_string_to_audfmt (var, defval, defaultp);
}
static int audio_get_conf_int (const char *key, int defval, int *defaultp)
{
int val;
char *strval;
strval = getenv (key);
if (strval) {
*defaultp = 0;
val = atoi (strval);
return val;
}
else {
*defaultp = 1;
return defval;
}
}
static const char *audio_get_conf_str (const char *key,
const char *defval,
int *defaultp)
{
const char *val = getenv (key);
if (!val) {
*defaultp = 1;
return defval;
}
else {
*defaultp = 0;
return val;
}
}
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (conf.log_to_monitor) {
if (cap) {
monitor_printf(cur_mon, "%s: ", cap);
}
monitor_vprintf(cur_mon, fmt, ap);
}
else {
if (cap) {
fprintf (stderr, "%s: ", cap);
}
vfprintf (stderr, fmt, ap);
}
}
void AUD_log (const char *cap, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (cap, fmt, ap);
va_end (ap);
}
static void audio_print_options (const char *prefix,
struct audio_option *opt)
{
char *uprefix;
if (!prefix) {
dolog ("No prefix specified\n");
return;
}
if (!opt) {
dolog ("No options\n");
return;
}
uprefix = audio_alloc_prefix (prefix);
for (; opt->name; opt++) {
const char *state = "default";
printf (" %s_%s: ", uprefix, opt->name);
if (opt->overriddenp && *opt->overriddenp) {
state = "current";
}
switch (opt->tag) {
case AUD_OPT_BOOL:
{
int *intp = opt->valp;
printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
}
break;
case AUD_OPT_INT:
{
int *intp = opt->valp;
printf ("integer, %s = %d\n", state, *intp);
}
break;
case AUD_OPT_FMT:
{
audfmt_e *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
audio_audfmt_to_string (*fmtp)
);
}
break;
case AUD_OPT_STR:
{
const char **strp = opt->valp;
printf ("string, %s = %s\n",
state,
*strp ? *strp : "(not set)");
}
break;
default:
printf ("???\n");
dolog ("Bad value tag for option %s_%s %d\n",
uprefix, opt->name, opt->tag);
break;
}
printf (" %s\n", opt->descr);
}
qemu_free (uprefix);
}
static void audio_process_options (const char *prefix,
struct audio_option *opt)
{
char *optname;
const char qemu_prefix[] = "QEMU_";
size_t preflen, optlen;
if (audio_bug (AUDIO_FUNC, !prefix)) {
dolog ("prefix = NULL\n");
return;
}
if (audio_bug (AUDIO_FUNC, !opt)) {
dolog ("opt = NULL\n");
return;
}
preflen = strlen (prefix);
for (; opt->name; opt++) {
size_t len, i;
int def;
if (!opt->valp) {
dolog ("Option value pointer for `%s' is not set\n",
opt->name);
continue;
}
len = strlen (opt->name);
/* len of opt->name + len of prefix + size of qemu_prefix
* (includes trailing zero) + zero + underscore (on behalf of
* sizeof) */
optlen = len + preflen + sizeof (qemu_prefix) + 1;
optname = qemu_malloc (optlen);
pstrcpy (optname, optlen, qemu_prefix);
/* copy while upper-casing, including trailing zero */
for (i = 0; i <= preflen; ++i) {
optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
}
pstrcat (optname, optlen, "_");
pstrcat (optname, optlen, opt->name);
def = 1;
switch (opt->tag) {
case AUD_OPT_BOOL:
case AUD_OPT_INT:
{
int *intp = opt->valp;
*intp = audio_get_conf_int (optname, *intp, &def);
}
break;
case AUD_OPT_FMT:
{
audfmt_e *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
case AUD_OPT_STR:
{
const char **strp = opt->valp;
*strp = audio_get_conf_str (optname, *strp, &def);
}
break;
default:
dolog ("Bad value tag for option `%s' - %d\n",
optname, opt->tag);
break;
}
if (!opt->overriddenp) {
opt->overriddenp = &opt->overridden;
}
*opt->overriddenp = !def;
qemu_free (optname);
}
}
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
case AUD_FMT_S8:
AUD_log (NULL, "S8");
break;
case AUD_FMT_U8:
AUD_log (NULL, "U8");
break;
case AUD_FMT_S16:
AUD_log (NULL, "S16");
break;
case AUD_FMT_U16:
AUD_log (NULL, "U16");
break;
case AUD_FMT_S32:
AUD_log (NULL, "S32");
break;
case AUD_FMT_U32:
AUD_log (NULL, "U32");
break;
default:
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
}
AUD_log (NULL, " endianness=");
switch (as->endianness) {
case 0:
AUD_log (NULL, "little");
break;
case 1:
AUD_log (NULL, "big");
break;
default:
AUD_log (NULL, "invalid");
break;
}
AUD_log (NULL, "\n");
}
static int audio_validate_settings (struct audsettings *as)
{
int invalid;
invalid = as->nchannels != 1 && as->nchannels != 2;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
case AUD_FMT_S16:
case AUD_FMT_U16:
case AUD_FMT_S32:
case AUD_FMT_U32:
break;
default:
invalid = 1;
break;
}
invalid |= as->freq <= 0;
return invalid ? -1 : 0;
}
static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0;
switch (as->fmt) {
case AUD_FMT_S8:
sign = 1;
case AUD_FMT_U8:
break;
case AUD_FMT_S16:
sign = 1;
case AUD_FMT_U16:
bits = 16;
break;
case AUD_FMT_S32:
sign = 1;
case AUD_FMT_U32:
bits = 32;
break;
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->sign == sign
&& info->bits == bits
&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
case AUD_FMT_S8:
sign = 1;
case AUD_FMT_U8:
break;
case AUD_FMT_S16:
sign = 1;
case AUD_FMT_U16:
bits = 16;
shift = 1;
break;
case AUD_FMT_S32:
sign = 1;
case AUD_FMT_U32:
bits = 32;
shift = 2;
break;
}
info->freq = as->freq;
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
info->shift = (as->nchannels == 2) + shift;
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
if (!len) {
return;
}
if (info->sign) {
memset (buf, 0x00, len << info->shift);
}
else {
switch (info->bits) {
case 8:
memset (buf, 0x80, len << info->shift);
break;
case 16:
{
int i;
uint16_t *p = buf;
int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
case 32:
{
int i;
uint32_t *p = buf;
int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
default:
AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
info->bits);
break;
}
}
}
/*
* Capture
*/
static void noop_conv (struct st_sample *dst, const void *src,
int samples, struct mixeng_volume *vol)
{
(void) src;
(void) dst;
(void) samples;
(void) vol;
}
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;
}
LIST_REMOVE (sw, entries);
LIST_REMOVE (sc, entries);
qemu_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 (AudioState *s, HWVoiceOut *hw)
{
CaptureVoiceOut *cap;
audio_detach_capture (hw);
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
SWVoiceCap *sc;
SWVoiceOut *sw;
HWVoiceOut *hw_cap = &cap->hw;
sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
if (!sc) {
dolog ("Could not allocate soft capture voice (%zu bytes)\n",
sizeof (*sc));
return -1;
}
sc->cap = cap;
sw = &sc->sw;
sw->hw = hw_cap;
sw->info = hw->info;
sw->empty = 1;
sw->active = hw->enabled;
sw->conv = noop_conv;
sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
sw->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));
qemu_free (sw);
return -1;
}
LIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
LIST_INSERT_HEAD (&hw->cap_head, sc, entries);
#ifdef DEBUG_CAPTURE
asprintf (&sw->name, "for %p %d,%d,%d",
hw, sw->info.freq, sw->info.bits, sw->info.nchannels);
dolog ("Added %s active = %d\n", sw->name, sw->active);
#endif
if (sw->active) {
audio_capture_maybe_changed (cap, 1);
}
}
return 0;
}
/*
* Hard voice (capture)
*/
static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
int m = hw->total_samples_captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
m = audio_MIN (m, sw->total_hw_samples_acquired);
}
}
return m;
}
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
{
int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
/*
* Soft voice (capture)
*/
static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
{
HWVoiceIn *hw = sw->hw;
int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
int rpos;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
rpos = hw->wpos - live;
if (rpos >= 0) {
return rpos;
}
else {
return hw->samples + rpos;
}
}
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
{
HWVoiceIn *hw = sw->hw;
int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
samples = size >> sw->info.shift;
if (!live) {
return 0;
}
swlim = (live * sw->ratio) >> 32;
swlim = audio_MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf + rpos;
isamp = hw->wpos - rpos;
/* XXX: <= ? */
if (isamp <= 0) {
isamp = hw->samples - rpos;
}
if (!isamp) {
break;
}
osamp = swlim;
if (audio_bug (AUDIO_FUNC, osamp < 0)) {
dolog ("osamp=%d\n", osamp);
return 0;
}
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->samples;
dst += osamp;
ret += osamp;
total += isamp;
}
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;
}
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
int smin;
smin = audio_pcm_hw_find_min_out (hw, nb_live);
if (!*nb_live) {
return 0;
}
else {
int live = smin;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
}
int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
{
int nb_live;
int live;
live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
return 0;
}
return live;
}
/*
* Soft voice (playback)
*/
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
{
int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
int ret = 0, pos = 0, total = 0;
if (!sw) {
return size;
}
hwsamples = sw->hw->samples;
live = sw->total_hw_samples_mixed;
if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
dolog ("live=%d hw->samples=%d\n", live, hwsamples);
return 0;
}
if (live == hwsamples) {
#ifdef DEBUG_OUT
dolog ("%s is full %d\n", sw->name, live);
#endif
return 0;
}
wpos = (sw->hw->rpos + live) % hwsamples;
samples = size >> sw->info.shift;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
swlim = audio_MIN (swlim, samples);
if (swlim) {
sw->conv (sw->buf, buf, swlim, &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"
int AUD_write (SWVoiceOut *sw, void *buf, int size)
{
int bytes;
if (!sw) {
/* XXX: Consider options */
return size;
}
if (!sw->hw->enabled) {
dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
bytes = sw->hw->pcm_ops->write (sw, buf, size);
return bytes;
}
int AUD_read (SWVoiceIn *sw, void *buf, int size)
{
int bytes;
if (!sw) {
/* XXX: Consider options */
return size;
}
if (!sw->hw->enabled) {
dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
bytes = sw->hw->pcm_ops->read (sw, buf, size);
return bytes;
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
return sw->hw->samples << sw->hw->info.shift;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
{
HWVoiceOut *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioState *s = &glob_audio_state;
SWVoiceOut *temp_sw;
SWVoiceCap *sc;
if (on) {
hw->pending_disable = 0;
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
}
}
}
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);
}
}
sw->total_hw_samples_acquired = hw->total_samples_captured;
}
else {
if (hw->enabled) {
int nb_active = 0;
for (temp_sw = hw->sw_head.lh_first; temp_sw;
temp_sw = temp_sw->entries.le_next) {
nb_active += temp_sw->active != 0;
}
if (nb_active == 1) {
hw->enabled = 0;
hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
}
}
}
sw->active = on;
}
}
static int audio_get_avail (SWVoiceIn *sw)
{
int live;
if (!sw) {
return 0;
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
return 0;
}
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
);
return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
}
static int audio_get_free (SWVoiceOut *sw)
{
int live, dead;
if (!sw) {
return 0;
}
live = sw->total_hw_samples_mixed;
if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
return 0;
}
dead = sw->hw->samples - live;
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
SW_NAME (sw),
live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
#endif
return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
}
static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
{
int n;
if (hw->enabled) {
SWVoiceCap *sc;
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
SWVoiceOut *sw = &sc->sw;
int rpos2 = rpos;
n = samples;
while (n) {
int till_end_of_hw = hw->samples - rpos2;
int to_write = audio_MIN (till_end_of_hw, n);
int bytes = to_write << hw->info.shift;
int written;
sw->buf = hw->mix_buf + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
dolog ("Could not mix %d bytes into a capture "
"buffer, mixed %d\n",
bytes, written);
break;
}
n -= to_write;
rpos2 = (rpos2 + to_write) % hw->samples;
}
}
}
n = audio_MIN (samples, hw->samples - rpos);
mixeng_clear (hw->mix_buf + rpos, n);
mixeng_clear (hw->mix_buf, samples - n);
}
static void audio_run_out (AudioState *s)
{
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
while ((hw = audio_pcm_hw_find_any_enabled_out (s, hw))) {
int played;
int live, free, nb_live, cleanup_required, prev_rpos;
live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
if (!nb_live) {
live = 0;
}
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
continue;
}
if (hw->pending_disable && !nb_live) {
SWVoiceCap *sc;
#ifdef DEBUG_OUT
dolog ("Disabling voice\n");
#endif
hw->enabled = 0;
hw->pending_disable = 0;
hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = 0;
audio_recalc_and_notify_capture (sc->cap);
}
continue;
}
if (!live) {
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
free = audio_get_free (sw);
if (free > 0) {
sw->callback.fn (sw->callback.opaque, free);
}
}
}
continue;
}
prev_rpos = hw->rpos;
played = hw->pcm_ops->run_out (hw);
if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
hw->rpos, hw->samples, played);
hw->rpos = 0;
}
#ifdef DEBUG_OUT
dolog ("played=%d\n", played);
#endif
if (played) {
hw->ts_helper += played;
audio_capture_mix_and_clear (hw, prev_rpos, played);
}
cleanup_required = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
continue;
}
if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
played, sw->total_hw_samples_mixed);
played = sw->total_hw_samples_mixed;
}
sw->total_hw_samples_mixed -= played;
if (!sw->total_hw_samples_mixed) {
sw->empty = 1;
cleanup_required |= !sw->active && !sw->callback.fn;
}
if (sw->active) {
free = audio_get_free (sw);
if (free > 0) {
sw->callback.fn (sw->callback.opaque, free);
}
}
}
if (cleanup_required) {
SWVoiceOut *sw1;
sw = hw->sw_head.lh_first;
while (sw) {
sw1 = sw->entries.le_next;
if (!sw->active && !sw->callback.fn) {
#ifdef DEBUG_PLIVE
dolog ("Finishing with old voice\n");
#endif
audio_close_out (s, sw);
}
sw = sw1;
}
}
}
}
static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
while ((hw = audio_pcm_hw_find_any_enabled_in (s, hw))) {
SWVoiceIn *sw;
int captured, min;
captured = hw->pcm_ops->run_in (hw);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
hw->ts_helper += captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
sw->total_hw_samples_acquired -= min;
if (sw->active) {
int avail;
avail = audio_get_avail (sw);
if (avail > 0) {
sw->callback.fn (sw->callback.opaque, avail);
}
}
}
}
}
static void audio_run_capture (AudioState *s)
{
CaptureVoiceOut *cap;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
int live, rpos, captured;
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw);
rpos = hw->rpos;
while (live) {
int left = hw->samples - rpos;
int to_capture = audio_MIN (live, left);
struct st_sample *src;
struct capture_callback *cb;
src = hw->mix_buf + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.capture (cb->opaque, cap->buf,
to_capture << hw->info.shift);
}
rpos = (rpos + to_capture) % hw->samples;
live -= to_capture;
}
hw->rpos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
continue;
}
if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
captured, sw->total_hw_samples_mixed);
captured = sw->total_hw_samples_mixed;
}
sw->total_hw_samples_mixed -= captured;
sw->empty = sw->total_hw_samples_mixed == 0;
}
}
}
static void audio_timer (void *opaque)
{
AudioState *s = opaque;
audio_run_out (s);
audio_run_in (s);
audio_run_capture (s);
qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
}
static struct audio_option audio_options[] = {
/* DAC */
{"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_out.enabled,
"Use fixed settings for host DAC", NULL, 0},
{"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
"Frequency for fixed host DAC", NULL, 0},
{"DAC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_out.settings.fmt,
"Format for fixed host DAC", NULL, 0},
{"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_out.settings.nchannels,
"Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},
{"DAC_VOICES", AUD_OPT_INT, &conf.fixed_out.nb_voices,
"Number of voices for DAC", NULL, 0},
/* ADC */
{"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_in.enabled,
"Use fixed settings for host ADC", NULL, 0},
{"ADC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_in.settings.freq,
"Frequency for fixed host ADC", NULL, 0},
{"ADC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_in.settings.fmt,
"Format for fixed host ADC", NULL, 0},
{"ADC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_in.settings.nchannels,
"Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},
{"ADC_VOICES", AUD_OPT_INT, &conf.fixed_in.nb_voices,
"Number of voices for ADC", NULL, 0},
/* Misc */
{"TIMER_PERIOD", AUD_OPT_INT, &conf.period.hertz,
"Timer period in HZ (0 - use lowest possible)", NULL, 0},
{"PLIVE", AUD_OPT_BOOL, &conf.plive,
"(undocumented)", NULL, 0},
{"LOG_TO_MONITOR", AUD_OPT_BOOL, &conf.log_to_monitor,
"print logging messages to monitor instead of stderr", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
};
static void audio_pp_nb_voices (const char *typ, int nb)
{
switch (nb) {
case 0:
printf ("Does not support %s\n", typ);
break;
case 1:
printf ("One %s voice\n", typ);
break;
case INT_MAX:
printf ("Theoretically supports many %s voices\n", typ);
break;
default:
printf ("Theoretically supports upto %d %s voices\n", nb, typ);
break;
}
}
void AUD_help (void)
{
size_t i;
audio_process_options ("AUDIO", audio_options);
for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
struct audio_driver *d = drvtab[i];
if (d->options) {
audio_process_options (d->name, d->options);
}
}
printf ("Audio options:\n");
audio_print_options ("AUDIO", audio_options);
printf ("\n");
printf ("Available drivers:\n");
for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
struct audio_driver *d = drvtab[i];
printf ("Name: %s\n", d->name);
printf ("Description: %s\n", d->descr);
audio_pp_nb_voices ("playback", d->max_voices_out);
audio_pp_nb_voices ("capture", d->max_voices_in);
if (d->options) {
printf ("Options:\n");
audio_print_options (d->name, d->options);
}
else {
printf ("No options\n");
}
printf ("\n");
}
printf (
"Options are settable through environment variables.\n"
"Example:\n"
#ifdef _WIN32
" set QEMU_AUDIO_DRV=wav\n"
" set QEMU_WAV_PATH=c:\\tune.wav\n"
#else
" export QEMU_AUDIO_DRV=wav\n"
" export QEMU_WAV_PATH=$HOME/tune.wav\n"
"(for csh replace export with setenv in the above)\n"
#endif
" qemu ...\n\n"
);
}
static int audio_driver_init (AudioState *s, struct audio_driver *drv)
{
if (drv->options) {
audio_process_options (drv->name, drv->options);
}
s->drv_opaque = drv->init ();
if (s->drv_opaque) {
audio_init_nb_voices_out (s, drv);
audio_init_nb_voices_in (s, drv);
s->drv = drv;
return 0;
}
else {
dolog ("Could not init `%s' audio driver\n", drv->name);
return -1;
}
}
static void audio_vm_change_state_handler (void *opaque, int running,
int reason)
{
AudioState *s = opaque;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
int op = running ? VOICE_ENABLE : VOICE_DISABLE;
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
hwo->pcm_ops->ctl_out (hwo, op);
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
hwi->pcm_ops->ctl_in (hwi, op);
}
}
static void audio_atexit (void)
{
AudioState *s = &glob_audio_state;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
SWVoiceCap *sc;
hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
hwo->pcm_ops->fini_out (hwo);
for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
CaptureVoiceOut *cap = sc->cap;
struct capture_callback *cb;
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.destroy (cb->opaque);
}
}
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
hwi->pcm_ops->fini_in (hwi);
}
if (s->drv) {
s->drv->fini (s->drv_opaque);
}
}
static void audio_save (QEMUFile *f, void *opaque)
{
(void) f;
(void) opaque;
}
static int audio_load (QEMUFile *f, void *opaque, int version_id)
{
(void) f;
(void) opaque;
if (version_id != 1) {
return -EINVAL;
}
return 0;
}
void AUD_register_card (AudioState *s, const char *name, QEMUSoundCard *card)
{
card->audio = s;
card->name = qemu_strdup (name);
memset (&card->entries, 0, sizeof (card->entries));
LIST_INSERT_HEAD (&s->card_head, card, entries);
}
void AUD_remove_card (QEMUSoundCard *card)
{
LIST_REMOVE (card, entries);
card->audio = NULL;
qemu_free (card->name);
}
AudioState *AUD_init (void)
{
size_t i;
int done = 0;
const char *drvname;
AudioState *s = &glob_audio_state;
LIST_INIT (&s->hw_head_out);
LIST_INIT (&s->hw_head_in);
LIST_INIT (&s->cap_head);
atexit (audio_atexit);
s->ts = qemu_new_timer (vm_clock, audio_timer, s);
if (!s->ts) {
dolog ("Could not create audio timer\n");
return NULL;
}
audio_process_options ("AUDIO", audio_options);
s->nb_hw_voices_out = conf.fixed_out.nb_voices;
s->nb_hw_voices_in = conf.fixed_in.nb_voices;
if (s->nb_hw_voices_out <= 0) {
dolog ("Bogus number of playback voices %d, setting to 1\n",
s->nb_hw_voices_out);
s->nb_hw_voices_out = 1;
}
if (s->nb_hw_voices_in <= 0) {
dolog ("Bogus number of capture voices %d, setting to 0\n",
s->nb_hw_voices_in);
s->nb_hw_voices_in = 0;
}
{
int def;
drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
}
if (drvname) {
int found = 0;
for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
if (!strcmp (drvname, drvtab[i]->name)) {
done = !audio_driver_init (s, drvtab[i]);
found = 1;
break;
}
}
if (!found) {
dolog ("Unknown audio driver `%s'\n", drvname);
dolog ("Run with -audio-help to list available drivers\n");
}
}
if (!done) {
for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
if (drvtab[i]->can_be_default) {
done = !audio_driver_init (s, drvtab[i]);
}
}
}
if (!done) {
done = !audio_driver_init (s, &no_audio_driver);
if (!done) {
dolog ("Could not initialize audio subsystem\n");
}
else {
dolog ("warning: Using timer based audio emulation\n");
}
}
if (done) {
VMChangeStateEntry *e;
if (conf.period.hertz <= 0) {
if (conf.period.hertz < 0) {
dolog ("warning: Timer period is negative - %d "
"treating as zero\n",
conf.period.hertz);
}
conf.period.ticks = 1;
}
else {
conf.period.ticks = ticks_per_sec / conf.period.hertz;
}
e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
if (!e) {
dolog ("warning: Could not register change state handler\n"
"(Audio can continue looping even after stopping the VM)\n");
}
}
else {
qemu_del_timer (s->ts);
return NULL;
}
LIST_INIT (&s->card_head);
register_savevm ("audio", 0, 1, audio_save, audio_load, s);
qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
return s;
}
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) {
/* XXX suppress */
s = &glob_audio_state;
}
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
goto err0;
}
cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
if (!cb) {
dolog ("Could not allocate capture callback information, size %zu\n",
sizeof (*cb));
goto err0;
}
cb->ops = *ops;
cb->opaque = cb_opaque;
cap = audio_pcm_capture_find_specific (s, as);
if (cap) {
LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
return cap;
}
else {
HWVoiceOut *hw;
CaptureVoiceOut *cap;
cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
if (!cap) {
dolog ("Could not allocate capture voice, size %zu\n",
sizeof (*cap));
goto err1;
}
hw = &cap->hw;
LIST_INIT (&hw->sw_head);
LIST_INIT (&cap->cb_head);
/* XXX find a more elegant way */
hw->samples = 4096 * 4;
hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
sizeof (struct st_sample));
if (!hw->mix_buf) {
dolog ("Could not allocate capture mix buffer (%d samples)\n",
hw->samples);
goto err2;
}
audio_pcm_init_info (&hw->info, as);
cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!cap->buf) {
dolog ("Could not allocate capture buffer "
"(%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
goto err3;
}
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
[hw->info.sign]
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
LIST_INSERT_HEAD (&s->cap_head, cap, entries);
LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
hw = NULL;
while ((hw = audio_pcm_hw_find_any_out (s, hw))) {
audio_attach_capture (s, hw);
}
return cap;
err3:
qemu_free (cap->hw.mix_buf);
err2:
qemu_free (cap);
err1:
qemu_free (cb);
err0:
return NULL;
}
}
void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
{
struct capture_callback *cb;
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
if (cb->opaque == cb_opaque) {
cb->ops.destroy (cb_opaque);
LIST_REMOVE (cb, entries);
qemu_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;
}
LIST_REMOVE (sw, entries);
LIST_REMOVE (sc, entries);
qemu_free (sc);
sw = sw1;
}
LIST_REMOVE (cap, entries);
qemu_free (cap);
}
return;
}
}
}
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
{
if (sw) {
sw->vol.mute = mute;
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
}
}
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
{
if (sw) {
sw->vol.mute = mute;
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
}
}
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