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qemu/audio/audio.c
Pavel Dovgalyuk 1ffc266539 audio: make audio poll timer deterministic 
This patch changes resetting strategy of the audio polling timer.
It does not change expiration time if the timer is already set.
This patch is needed to make this timer deterministic and to use execution
record/replay for audio devices.

audio_reset_timer is used in the function audio_vm_change_state_handler.
Therefore every time VM is stopped or restarted the timer will be reset
to new timeout. Virtual clock does not proceed while VM is stopped.
Therefore there is no need in resetting the timeout when VM restarts.

v2: updated commit message
v3: now using timer_mod_anticipate function (as suggested by Yurii Zubrytskyi)

Signed-off-by: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>
Message-id: 20170214071510.6112.76764.stgit@PASHA-ISP
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2017-03-01 15:12:03 +01:00

2066 lines
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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 "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
#include "qemu/timer.h"
#include "sysemu/sysemu.h"
#include "qemu/cutils.h"
#include "sysemu/replay.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.
*/
static struct audio_driver *drvtab[] = {
#ifdef CONFIG_SPICE
&spice_audio_driver,
#endif
CONFIG_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 try_poll_in;
int try_poll_out;
} conf = {
.fixed_out = { /* DAC fixed settings */
.enabled = 1,
.nb_voices = 1,
.greedy = 1,
.settings = {
.freq = 44100,
.nchannels = 2,
.fmt = AUD_FMT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
.fixed_in = { /* ADC fixed settings */
.enabled = 1,
.nb_voices = 1,
.greedy = 1,
.settings = {
.freq = 44100,
.nchannels = 2,
.fmt = AUD_FMT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
.period = { .hertz = 100 },
.try_poll_in = 1,
.try_poll_out = 1,
};
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
static void audio_print_options (const char *prefix,
struct audio_option *opt);
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) {
struct audio_driver *d;
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
AUD_log (NULL, "Please send bug report to av1474@comtv.ru\n");
AUD_log (NULL, "I am sorry\n");
d = glob_audio_state.drv;
if (d) {
audio_print_options (d->name, d->options);
}
}
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);
}
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 = g_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 (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);
}
g_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 = g_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;
g_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;
/* fall through */
case AUD_FMT_U8:
break;
case AUD_FMT_S16:
sign = 1;
/* fall through */
case AUD_FMT_U16:
bits = 16;
break;
case AUD_FMT_S32:
sign = 1;
/* fall through */
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)
{
(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 = 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->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 (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_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 (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;
}
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 (AUDIO_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 (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);
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 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) + conf.period.ticks);
}
else {
timer_del (s->ts);
}
}
static void audio_timer (void *opaque)
{
audio_run ("timer");
audio_reset_timer (opaque);
}
/*
* 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, conf.try_poll_out);
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, conf.try_poll_in);
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 (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 (hw))) {
int played;
int live, free, nb_live, cleanup_required, prev_rpos;
live = audio_pcm_hw_get_live_out (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, live);
replay_audio_out(&played);
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) {
audio_close_out (sw);
}
sw = sw1;
}
}
}
}
static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
SWVoiceIn *sw;
int captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
captured = hw->pcm_ops->run_in(hw);
}
replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
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, NULL);
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;
}
}
}
void audio_run (const char *msg)
{
AudioState *s = &glob_audio_state;
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
}
static struct audio_option audio_options[] = {
/* DAC */
{
.name = "DAC_FIXED_SETTINGS",
.tag = AUD_OPT_BOOL,
.valp = &conf.fixed_out.enabled,
.descr = "Use fixed settings for host DAC"
},
{
.name = "DAC_FIXED_FREQ",
.tag = AUD_OPT_INT,
.valp = &conf.fixed_out.settings.freq,
.descr = "Frequency for fixed host DAC"
},
{
.name = "DAC_FIXED_FMT",
.tag = AUD_OPT_FMT,
.valp = &conf.fixed_out.settings.fmt,
.descr = "Format for fixed host DAC"
},
{
.name = "DAC_FIXED_CHANNELS",
.tag = AUD_OPT_INT,
.valp = &conf.fixed_out.settings.nchannels,
.descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
},
{
.name = "DAC_VOICES",
.tag = AUD_OPT_INT,
.valp = &conf.fixed_out.nb_voices,
.descr = "Number of voices for DAC"
},
{
.name = "DAC_TRY_POLL",
.tag = AUD_OPT_BOOL,
.valp = &conf.try_poll_out,
.descr = "Attempt using poll mode for DAC"
},
/* ADC */
{
.name = "ADC_FIXED_SETTINGS",
.tag = AUD_OPT_BOOL,
.valp = &conf.fixed_in.enabled,
.descr = "Use fixed settings for host ADC"
},
{
.name = "ADC_FIXED_FREQ",
.tag = AUD_OPT_INT,
.valp = &conf.fixed_in.settings.freq,
.descr = "Frequency for fixed host ADC"
},
{
.name = "ADC_FIXED_FMT",
.tag = AUD_OPT_FMT,
.valp = &conf.fixed_in.settings.fmt,
.descr = "Format for fixed host ADC"
},
{
.name = "ADC_FIXED_CHANNELS",
.tag = AUD_OPT_INT,
.valp = &conf.fixed_in.settings.nchannels,
.descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
},
{
.name = "ADC_VOICES",
.tag = AUD_OPT_INT,
.valp = &conf.fixed_in.nb_voices,
.descr = "Number of voices for ADC"
},
{
.name = "ADC_TRY_POLL",
.tag = AUD_OPT_BOOL,
.valp = &conf.try_poll_in,
.descr = "Attempt using poll mode for ADC"
},
/* Misc */
{
.name = "TIMER_PERIOD",
.tag = AUD_OPT_INT,
.valp = &conf.period.hertz,
.descr = "Timer period in HZ (0 - use lowest possible)"
},
{ /* End of list */ }
};
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 up to %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 (drv);
audio_init_nb_voices_in (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,
RunState state)
{
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 (hwo))) {
hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
}
audio_reset_timer (s);
}
static bool is_cleaning_up;
bool audio_is_cleaning_up(void)
{
return is_cleaning_up;
}
void audio_cleanup(void)
{
AudioState *s = &glob_audio_state;
HWVoiceOut *hwo, *hwon;
HWVoiceIn *hwi, *hwin;
is_cleaning_up = true;
QLIST_FOREACH_SAFE(hwo, &glob_audio_state.hw_head_out, entries, hwon) {
SWVoiceCap *sc;
if (hwo->enabled) {
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);
}
}
QLIST_REMOVE(hwo, entries);
}
QLIST_FOREACH_SAFE(hwi, &glob_audio_state.hw_head_in, entries, hwin) {
if (hwi->enabled) {
hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
}
hwi->pcm_ops->fini_in (hwi);
QLIST_REMOVE(hwi, entries);
}
if (s->drv) {
s->drv->fini (s->drv_opaque);
s->drv = NULL;
}
}
static const VMStateDescription vmstate_audio = {
.name = "audio",
.version_id = 1,
.minimum_version_id = 1,
.fields = (VMStateField[]) {
VMSTATE_END_OF_LIST()
}
};
static void audio_init (void)
{
size_t i;
int done = 0;
const char *drvname;
VMChangeStateEntry *e;
AudioState *s = &glob_audio_state;
if (s->drv) {
return;
}
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
atexit(audio_cleanup);
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
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);
assert(done);
dolog("warning: Using timer based audio emulation\n");
}
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 = NANOSECONDS_PER_SECOND / 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");
}
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
}
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
audio_init ();
card->name = g_strdup (name);
memset (&card->entries, 0, sizeof (card->entries));
QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
}
void AUD_remove_card (QEMUSoundCard *card)
{
QLIST_REMOVE (card, entries);
g_free (card->name);
}
CaptureVoiceOut *AUD_add_capture (
struct audsettings *as,
struct audio_capture_ops *ops,
void *cb_opaque
)
{
AudioState *s = &glob_audio_state;
CaptureVoiceOut *cap;
struct capture_callback *cb;
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 (as);
if (cap) {
QLIST_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;
QLIST_INIT (&hw->sw_head);
QLIST_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)];
QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
QLIST_FOREACH(hw, &glob_audio_state.hw_head_out, entries) {
audio_attach_capture (hw);
}
return cap;
err3:
g_free (cap->hw.mix_buf);
err2:
g_free (cap);
err1:
g_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);
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);
}
return;
}
}
}
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
{
if (sw) {
HWVoiceOut *hw = sw->hw;
sw->vol.mute = mute;
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
if (hw->pcm_ops->ctl_out) {
hw->pcm_ops->ctl_out (hw, VOICE_VOLUME, sw);
}
}
}
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
{
if (sw) {
HWVoiceIn *hw = sw->hw;
sw->vol.mute = mute;
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
if (hw->pcm_ops->ctl_in) {
hw->pcm_ops->ctl_in (hw, VOICE_VOLUME, sw);
}
}
}
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