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audio.c
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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 "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
#include "qemu/timer.h"
#include "sysemu/sysemu.h"
#include "qemu/cutils.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 [email protected]\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;
}