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FRET-qemu/audio/audio.c
Bruce Rogers cbaf25d1f5 audio: fix wavcapture segfault 
Commit 571a8c522e caused the HMP wavcapture command to segfault when
processing audio data in audio_pcm_sw_write(), where a NULL
sw->hw->pcm_ops is dereferenced. This fix checks that the pointer is
valid before dereferincing it. A similar fix is also made in the
parallel function audio_pcm_sw_read().

Fixes: 571a8c522e (audio: split ctl_* functions into enable_* and
volume_*)
Signed-off-by: Bruce Rogers <brogers@suse.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20200521172931.121903-1-brogers@suse.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-05-26 07:55:23 +02:00

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/*
* QEMU Audio subsystem
*
* Copyright (c) 2003-2005 Vassili Karpov (malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "qemu/osdep.h"
#include "audio.h"
#include "migration/vmstate.h"
#include "monitor/monitor.h"
#include "qemu/timer.h"
#include "qapi/error.h"
#include "qapi/qobject-input-visitor.h"
#include "qapi/qapi-visit-audio.h"
#include "qemu/cutils.h"
#include "qemu/module.h"
#include "sysemu/replay.h"
#include "sysemu/runstate.h"
#include "trace.h"
#define AUDIO_CAP "audio"
#include "audio_int.h"
/* #define DEBUG_LIVE */
/* #define DEBUG_OUT */
/* #define DEBUG_CAPTURE */
/* #define DEBUG_POLL */
#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
/* Order of CONFIG_AUDIO_DRIVERS is import.
The 1st one is the one used by default, that is the reason
that we generate the list.
*/
const char *audio_prio_list[] = {
"spice",
CONFIG_AUDIO_DRIVERS
"none",
"wav",
NULL
};
static QLIST_HEAD(, audio_driver) audio_drivers;
static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
void audio_driver_register(audio_driver *drv)
{
QLIST_INSERT_HEAD(&audio_drivers, drv, next);
}
audio_driver *audio_driver_lookup(const char *name)
{
struct audio_driver *d;
QLIST_FOREACH(d, &audio_drivers, next) {
if (strcmp(name, d->name) == 0) {
return d;
}
}
audio_module_load_one(name);
QLIST_FOREACH(d, &audio_drivers, next) {
if (strcmp(name, d->name) == 0) {
return d;
}
}
return NULL;
}
static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
QTAILQ_HEAD_INITIALIZER(audio_states);
const struct mixeng_volume nominal_volume = {
.mute = 0,
#ifdef FLOAT_MIXENG
.r = 1.0,
.l = 1.0,
#else
.r = 1ULL << 32,
.l = 1ULL << 32,
#endif
};
static bool legacy_config = true;
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
int audio_bug (const char *funcname, int cond)
{
if (cond) {
static int shown;
AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
if (!shown) {
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
AUD_log (NULL, "I am sorry\n");
}
AUD_log (NULL, "Context:\n");
#if defined AUDIO_BREAKPOINT_ON_BUG
# if defined HOST_I386
# if defined __GNUC__
__asm__ ("int3");
# elif defined _MSC_VER
_asm _emit 0xcc;
# else
abort ();
# endif
# else
abort ();
# endif
#endif
}
return cond;
}
#endif
static inline int audio_bits_to_index (int bits)
{
switch (bits) {
case 8:
return 0;
case 16:
return 1;
case 32:
return 2;
default:
audio_bug ("bits_to_index", 1);
AUD_log (NULL, "invalid bits %d\n", bits);
return 0;
}
}
void *audio_calloc (const char *funcname, int nmemb, size_t size)
{
int cond;
size_t len;
len = nmemb * size;
cond = !nmemb || !size;
cond |= nmemb < 0;
cond |= len < size;
if (audio_bug ("audio_calloc", cond)) {
AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
funcname);
AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
return NULL;
}
return g_malloc0 (len);
}
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
fprintf(stderr, "%s: ", cap);
}
vfprintf(stderr, fmt, ap);
}
void AUD_log (const char *cap, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (cap, fmt, ap);
va_end (ap);
}
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
case AUDIO_FORMAT_S8:
AUD_log (NULL, "S8");
break;
case AUDIO_FORMAT_U8:
AUD_log (NULL, "U8");
break;
case AUDIO_FORMAT_S16:
AUD_log (NULL, "S16");
break;
case AUDIO_FORMAT_U16:
AUD_log (NULL, "U16");
break;
case AUDIO_FORMAT_S32:
AUD_log (NULL, "S32");
break;
case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
case AUDIO_FORMAT_F32:
AUD_log (NULL, "F32");
break;
default:
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
}
AUD_log (NULL, " endianness=");
switch (as->endianness) {
case 0:
AUD_log (NULL, "little");
break;
case 1:
AUD_log (NULL, "big");
break;
default:
AUD_log (NULL, "invalid");
break;
}
AUD_log (NULL, "\n");
}
static int audio_validate_settings (struct audsettings *as)
{
int invalid;
invalid = as->nchannels < 1;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
case AUDIO_FORMAT_U8:
case AUDIO_FORMAT_S16:
case AUDIO_FORMAT_U16:
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_U32:
case AUDIO_FORMAT_F32:
break;
default:
invalid = 1;
break;
}
invalid |= as->freq <= 0;
return invalid ? -1 : 0;
}
static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8;
bool is_signed = false, is_float = false;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
is_signed = true;
/* fall through */
case AUDIO_FORMAT_U8:
break;
case AUDIO_FORMAT_S16:
is_signed = true;
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
break;
case AUDIO_FORMAT_F32:
is_float = true;
/* fall through */
case AUDIO_FORMAT_S32:
is_signed = true;
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
break;
default:
abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->is_signed == is_signed
&& info->is_float == is_float
&& info->bits == bits
&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, mul;
bool is_signed = false, is_float = false;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
is_signed = true;
/* fall through */
case AUDIO_FORMAT_U8:
mul = 1;
break;
case AUDIO_FORMAT_S16:
is_signed = true;
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
mul = 2;
break;
case AUDIO_FORMAT_F32:
is_float = true;
/* fall through */
case AUDIO_FORMAT_S32:
is_signed = true;
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
mul = 4;
break;
default:
abort();
}
info->freq = as->freq;
info->bits = bits;
info->is_signed = is_signed;
info->is_float = is_float;
info->nchannels = as->nchannels;
info->bytes_per_frame = as->nchannels * mul;
info->bytes_per_second = info->freq * info->bytes_per_frame;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
if (!len) {
return;
}
if (info->is_signed || info->is_float) {
memset(buf, 0x00, len * info->bytes_per_frame);
}
else {
switch (info->bits) {
case 8:
memset(buf, 0x80, len * info->bytes_per_frame);
break;
case 16:
{
int i;
uint16_t *p = buf;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
break;
case 32:
{
int i;
uint32_t *p = buf;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
break;
default:
AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
info->bits);
break;
}
}
}
/*
* Capture
*/
static void noop_conv (struct st_sample *dst, const void *src, int samples)
{
(void) src;
(void) dst;
(void) samples;
}
static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
struct audsettings *as)
{
CaptureVoiceOut *cap;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
if (audio_pcm_info_eq (&cap->hw.info, as)) {
return cap;
}
}
return NULL;
}
static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
{
struct capture_callback *cb;
#ifdef DEBUG_CAPTURE
dolog ("notification %d sent\n", cmd);
#endif
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.notify (cb->opaque, cmd);
}
}
static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
if (cap->hw.enabled != enabled) {
audcnotification_e cmd;
cap->hw.enabled = enabled;
cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
audio_notify_capture (cap, cmd);
}
}
static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
{
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
int enabled = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
enabled = 1;
break;
}
}
audio_capture_maybe_changed (cap, enabled);
}
static void audio_detach_capture (HWVoiceOut *hw)
{
SWVoiceCap *sc = hw->cap_head.lh_first;
while (sc) {
SWVoiceCap *sc1 = sc->entries.le_next;
SWVoiceOut *sw = &sc->sw;
CaptureVoiceOut *cap = sc->cap;
int was_active = sw->active;
if (sw->rate) {
st_rate_stop (sw->rate);
sw->rate = NULL;
}
QLIST_REMOVE (sw, entries);
QLIST_REMOVE (sc, entries);
g_free (sc);
if (was_active) {
/* We have removed soft voice from the capture:
this might have changed the overall status of the capture
since this might have been the only active voice */
audio_recalc_and_notify_capture (cap);
}
sc = sc1;
}
}
static int audio_attach_capture (HWVoiceOut *hw)
{
AudioState *s = hw->s;
CaptureVoiceOut *cap;
audio_detach_capture (hw);
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
SWVoiceCap *sc;
SWVoiceOut *sw;
HWVoiceOut *hw_cap = &cap->hw;
sc = g_malloc0(sizeof(*sc));
sc->cap = cap;
sw = &sc->sw;
sw->hw = hw_cap;
sw->info = hw->info;
sw->empty = 1;
sw->active = hw->enabled;
sw->conv = noop_conv;
sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
sw->vol = nominal_volume;
sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
if (!sw->rate) {
dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
g_free (sw);
return -1;
}
QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
#ifdef DEBUG_CAPTURE
sw->name = g_strdup_printf ("for %p %d,%d,%d",
hw, sw->info.freq, sw->info.bits,
sw->info.nchannels);
dolog ("Added %s active = %d\n", sw->name, sw->active);
#endif
if (sw->active) {
audio_capture_maybe_changed (cap, 1);
}
}
return 0;
}
/*
* Hard voice (capture)
*/
static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
size_t m = hw->total_samples_captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
m = MIN (m, sw->total_hw_samples_acquired);
}
}
return m;
}
static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
if (audio_bug(__func__, live > hw->conv_buf->size)) {
dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
return live;
}
static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
size_t pos = hw->mix_buf->pos;
while (len) {
st_sample *src = hw->mix_buf->samples + pos;
uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
hw->clip(dst, src, samples_to_clip);
pos = (pos + samples_to_clip) % hw->mix_buf->size;
len -= samples_to_clip;
clipped += samples_to_clip;
}
}
/*
* Soft voice (capture)
*/
static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
{
HWVoiceIn *hw = sw->hw;
ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
ssize_t rpos;
if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
rpos = hw->conv_buf->pos - live;
if (rpos >= 0) {
return rpos;
}
else {
return hw->conv_buf->size + rpos;
}
}
static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
{
HWVoiceIn *hw = sw->hw;
size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug(__func__, live > hw->conv_buf->size)) {
dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
samples = size / sw->info.bytes_per_frame;
if (!live) {
return 0;
}
swlim = (live * sw->ratio) >> 32;
swlim = MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf->samples + rpos;
if (hw->conv_buf->pos > rpos) {
isamp = hw->conv_buf->pos - rpos;
} else {
isamp = hw->conv_buf->size - rpos;
}
if (!isamp) {
break;
}
osamp = swlim;
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->conv_buf->size;
dst += osamp;
ret += osamp;
total += isamp;
}
if (hw->pcm_ops && !hw->pcm_ops->volume_in) {
mixeng_volume (sw->buf, ret, &sw->vol);
}
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
return ret * sw->info.bytes_per_frame;
}
/*
* Hard voice (playback)
*/
static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
SWVoiceOut *sw;
size_t m = SIZE_MAX;
int nb_live = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active || !sw->empty) {
m = MIN (m, sw->total_hw_samples_mixed);
nb_live += 1;
}
}
*nb_livep = nb_live;
return m;
}
static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
{
size_t smin;
int nb_live1;
smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
if (nb_live) {
*nb_live = nb_live1;
}
if (nb_live1) {
size_t live = smin;
if (audio_bug(__func__, live > hw->mix_buf->size)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
return 0;
}
return live;
}
return 0;
}
/*
* Soft voice (playback)
*/
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
{
size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
size_t ret = 0, pos = 0, total = 0;
if (!sw) {
return size;
}
hwsamples = sw->hw->mix_buf->size;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hwsamples)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
return 0;
}
if (live == hwsamples) {
#ifdef DEBUG_OUT
dolog ("%s is full %d\n", sw->name, live);
#endif
return 0;
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
samples = size / sw->info.bytes_per_frame;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
swlim = MIN (swlim, samples);
if (swlim) {
sw->conv (sw->buf, buf, swlim);
if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) {
mixeng_volume (sw->buf, swlim, &sw->vol);
}
}
while (swlim) {
dead = hwsamples - live;
left = hwsamples - wpos;
blck = MIN (dead, left);
if (!blck) {
break;
}
isamp = swlim;
osamp = blck;
st_rate_flow_mix (
sw->rate,
sw->buf + pos,
sw->hw->mix_buf->samples + 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 %zu ret %zu total sw %zu\n",
SW_NAME (sw),
size / sw->info.bytes_per_frame,
ret,
sw->total_hw_samples_mixed
);
#endif
return ret * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
{
dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
cap, info->bits, info->is_signed, info->is_float, info->freq,
info->nchannels);
}
#endif
#define DAC
#include "audio_template.h"
#undef DAC
#include "audio_template.h"
/*
* Timer
*/
static int audio_is_timer_needed(AudioState *s)
{
HWVoiceIn *hwi = NULL;
HWVoiceOut *hwo = NULL;
while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
if (!hwo->poll_mode) return 1;
}
while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
if (!hwi->poll_mode) return 1;
}
return 0;
}
static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed(s)) {
timer_mod_anticipate_ns(s->ts,
qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
if (!s->timer_running) {
s->timer_running = true;
s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
if (s->timer_running) {
s->timer_running = false;
trace_audio_timer_stop();
}
}
}
static void audio_timer (void *opaque)
{
int64_t now, diff;
AudioState *s = opaque;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
diff = now - s->timer_last;
if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
s->timer_last = now;
audio_run(s, "timer");
audio_reset_timer(s);
}
/*
* Public API
*/
size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
{
HWVoiceOut *hw;
if (!sw) {
/* XXX: Consider options */
return size;
}
hw = sw->hw;
if (!hw->enabled) {
dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
return audio_pcm_sw_write(sw, buf, size);
} else {
return hw->pcm_ops->write(hw, buf, size);
}
}
size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
{
HWVoiceIn *hw;
if (!sw) {
/* XXX: Consider options */
return size;
}
hw = sw->hw;
if (!hw->enabled) {
dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
return audio_pcm_sw_read(sw, buf, size);
} else {
return hw->pcm_ops->read(hw, buf, size);
}
}
int AUD_get_buffer_size_out(SWVoiceOut *sw)
{
return sw->hw->samples * sw->hw->info.bytes_per_frame;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
{
HWVoiceOut *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioState *s = sw->s;
SWVoiceOut *temp_sw;
SWVoiceCap *sc;
if (on) {
hw->pending_disable = 0;
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
if (hw->pcm_ops->enable_out) {
hw->pcm_ops->enable_out(hw, true);
}
audio_reset_timer (s);
}
}
}
else {
if (hw->enabled) {
int nb_active = 0;
for (temp_sw = hw->sw_head.lh_first; temp_sw;
temp_sw = temp_sw->entries.le_next) {
nb_active += temp_sw->active != 0;
}
hw->pending_disable = nb_active == 1;
}
}
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = hw->enabled;
if (hw->enabled) {
audio_capture_maybe_changed (sc->cap, 1);
}
}
sw->active = on;
}
}
void AUD_set_active_in (SWVoiceIn *sw, int on)
{
HWVoiceIn *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioState *s = sw->s;
SWVoiceIn *temp_sw;
if (on) {
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
if (hw->pcm_ops->enable_in) {
hw->pcm_ops->enable_in(hw, true);
}
audio_reset_timer (s);
}
}
sw->total_hw_samples_acquired = hw->total_samples_captured;
}
else {
if (hw->enabled) {
int nb_active = 0;
for (temp_sw = hw->sw_head.lh_first; temp_sw;
temp_sw = temp_sw->entries.le_next) {
nb_active += temp_sw->active != 0;
}
if (nb_active == 1) {
hw->enabled = 0;
if (hw->pcm_ops->enable_in) {
hw->pcm_ops->enable_in(hw, false);
}
}
}
}
sw->active = on;
}
}
static size_t audio_get_avail (SWVoiceIn *sw)
{
size_t live;
if (!sw) {
return 0;
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
sw->hw->conv_buf->size);
return 0;
}
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
);
return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static size_t audio_get_free(SWVoiceOut *sw)
{
size_t live, dead;
if (!sw) {
return 0;
}
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
sw->hw->mix_buf->size);
return 0;
}
dead = sw->hw->mix_buf->size - live;
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
SW_NAME (sw),
live, dead, (((int64_t) dead << 32) / sw->ratio) *
sw->info.bytes_per_frame);
#endif
return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
size_t samples)
{
size_t n;
if (hw->enabled) {
SWVoiceCap *sc;
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
SWVoiceOut *sw = &sc->sw;
int rpos2 = rpos;
n = samples;
while (n) {
size_t till_end_of_hw = hw->mix_buf->size - rpos2;
size_t to_write = MIN(till_end_of_hw, n);
size_t bytes = to_write * hw->info.bytes_per_frame;
size_t written;
sw->buf = hw->mix_buf->samples + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
dolog("Could not mix %zu bytes into a capture "
"buffer, mixed %zu\n",
bytes, written);
break;
}
n -= to_write;
rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
}
}
}
n = MIN(samples, hw->mix_buf->size - rpos);
mixeng_clear(hw->mix_buf->samples + rpos, n);
mixeng_clear(hw->mix_buf->samples, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
{
size_t clipped = 0;
while (live) {
size_t size, decr, proc;
void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
if (!buf || size == 0) {
break;
}
decr = MIN(size / hw->info.bytes_per_frame, live);
audio_pcm_hw_clip_out(hw, buf, decr);
proc = hw->pcm_ops->put_buffer_out(hw, buf,
decr * hw->info.bytes_per_frame) /
hw->info.bytes_per_frame;
live -= proc;
clipped += proc;
hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
if (proc == 0 || proc < decr) {
break;
}
}
if (hw->pcm_ops->run_buffer_out) {
hw->pcm_ops->run_buffer_out(hw);
}
return clipped;
}
static void audio_run_out (AudioState *s)
{
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
if (!audio_get_pdo_out(s->dev)->mixing_engine) {
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
/* there is exactly 1 sw for each hw with no mixeng */
sw = hw->sw_head.lh_first;
if (hw->pending_disable) {
hw->enabled = 0;
hw->pending_disable = 0;
if (hw->pcm_ops->enable_out) {
hw->pcm_ops->enable_out(hw, false);
}
}
if (sw->active) {
sw->callback.fn(sw->callback.opaque, INT_MAX);
}
}
return;
}
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
size_t played, live, prev_rpos, free;
int nb_live, cleanup_required;
live = audio_pcm_hw_get_live_out (hw, &nb_live);
if (!nb_live) {
live = 0;
}
if (audio_bug(__func__, live > hw->mix_buf->size)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
continue;
}
if (hw->pending_disable && !nb_live) {
SWVoiceCap *sc;
#ifdef DEBUG_OUT
dolog ("Disabling voice\n");
#endif
hw->enabled = 0;
hw->pending_disable = 0;
if (hw->pcm_ops->enable_out) {
hw->pcm_ops->enable_out(hw, false);
}
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = 0;
audio_recalc_and_notify_capture (sc->cap);
}
continue;
}
if (!live) {
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->mix_buf->pos;
played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
hw->mix_buf->pos, hw->mix_buf->size, played);
hw->mix_buf->pos = 0;
}
#ifdef DEBUG_OUT
dolog("played=%zu\n", played);
#endif
if (played) {
hw->ts_helper += played;
audio_capture_mix_and_clear (hw, prev_rpos, played);
}
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(__func__, played > sw->total_hw_samples_mixed)) {
dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
played, sw->total_hw_samples_mixed);
played = sw->total_hw_samples_mixed;
}
sw->total_hw_samples_mixed -= played;
if (!sw->total_hw_samples_mixed) {
sw->empty = 1;
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 size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
{
size_t conv = 0;
STSampleBuffer *conv_buf = hw->conv_buf;
while (samples) {
size_t proc;
size_t size = samples * hw->info.bytes_per_frame;
void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
assert(size % hw->info.bytes_per_frame == 0);
if (size == 0) {
hw->pcm_ops->put_buffer_in(hw, buf, size);
break;
}
proc = MIN(size / hw->info.bytes_per_frame,
conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
}
return conv;
}
static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
if (!audio_get_pdo_in(s->dev)->mixing_engine) {
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
/* there is exactly 1 sw for each hw with no mixeng */
SWVoiceIn *sw = hw->sw_head.lh_first;
if (sw->active) {
sw->callback.fn(sw->callback.opaque, INT_MAX);
}
}
return;
}
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
captured = audio_pcm_hw_run_in(
hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
}
replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
hw->conv_buf->size);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
hw->ts_helper += captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
sw->total_hw_samples_acquired -= min;
if (sw->active) {
size_t 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) {
size_t live, rpos, captured;
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
rpos = hw->mix_buf->pos;
while (live) {
size_t left = hw->mix_buf->size - rpos;
size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
src = hw->mix_buf->samples + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.capture (cb->opaque, cap->buf,
to_capture * hw->info.bytes_per_frame);
}
rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
}
hw->mix_buf->pos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
continue;
}
if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
captured, sw->total_hw_samples_mixed);
captured = sw->total_hw_samples_mixed;
}
sw->total_hw_samples_mixed -= captured;
sw->empty = sw->total_hw_samples_mixed == 0;
}
}
}
void audio_run(AudioState *s, const char *msg)
{
audio_run_out(s);
audio_run_in(s);
audio_run_capture(s);
#ifdef DEBUG_POLL
{
static double prevtime;
double currtime;
struct timeval tv;
if (gettimeofday (&tv, NULL)) {
perror ("audio_run: gettimeofday");
return;
}
currtime = tv.tv_sec + tv.tv_usec * 1e-6;
dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
prevtime = currtime;
}
#endif
}
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
{
ssize_t start;
if (unlikely(!hw->buf_emul)) {
size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
}
while (hw->pending_emul < hw->size_emul) {
size_t read_len = MIN(hw->size_emul - hw->pos_emul,
hw->size_emul - hw->pending_emul);
size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
read_len);
hw->pending_emul += read;
hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
if (read < read_len) {
break;
}
}
start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
if (start < 0) {
start += hw->size_emul;
}
assert(start >= 0 && start < hw->size_emul);
*size = MIN(*size, hw->pending_emul);
*size = MIN(*size, hw->size_emul - start);
return hw->buf_emul + start;
}
void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
{
assert(size <= hw->pending_emul);
hw->pending_emul -= size;
}
void audio_generic_run_buffer_out(HWVoiceOut *hw)
{
while (hw->pending_emul) {
size_t write_len, written;
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
if (start < 0) {
start += hw->size_emul;
}
assert(start >= 0 && start < hw->size_emul);
write_len = MIN(hw->pending_emul, hw->size_emul - start);
written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
hw->pending_emul -= written;
if (written < write_len) {
break;
}
}
}
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
}
*size = MIN(hw->size_emul - hw->pending_emul,
hw->size_emul - hw->pos_emul);
return hw->buf_emul + hw->pos_emul;
}
size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
{
assert(buf == hw->buf_emul + hw->pos_emul &&
size + hw->pending_emul <= hw->size_emul);
hw->pending_emul += size;
hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
return size;
}
size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
{
size_t dst_size, copy_size;
void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
copy_size = MIN(size, dst_size);
memcpy(dst, buf, copy_size);
return hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
}
size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
{
void *src = hw->pcm_ops->get_buffer_in(hw, &size);
memcpy(buf, src, size);
hw->pcm_ops->put_buffer_in(hw, src, size);
return size;
}
static int audio_driver_init(AudioState *s, struct audio_driver *drv,
bool msg, Audiodev *dev)
{
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
if (!drv->pcm_ops->get_buffer_in) {
drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
}
if (!drv->pcm_ops->get_buffer_out) {
drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
}
audio_init_nb_voices_out(s, drv);
audio_init_nb_voices_in(s, drv);
s->drv = drv;
return 0;
}
else {
if (msg) {
dolog("Could not init `%s' audio driver\n", drv->name);
}
return -1;
}
}
static void audio_vm_change_state_handler (void *opaque, int running,
RunState state)
{
AudioState *s = opaque;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
if (hwo->pcm_ops->enable_out) {
hwo->pcm_ops->enable_out(hwo, running);
}
}
while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
if (hwi->pcm_ops->enable_in) {
hwi->pcm_ops->enable_in(hwi, running);
}
}
audio_reset_timer (s);
}
static bool is_cleaning_up;
bool audio_is_cleaning_up(void)
{
return is_cleaning_up;
}
static void free_audio_state(AudioState *s)
{
HWVoiceOut *hwo, *hwon;
HWVoiceIn *hwi, *hwin;
QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
if (hwo->enabled && hwo->pcm_ops->enable_out) {
hwo->pcm_ops->enable_out(hwo, false);
}
hwo->pcm_ops->fini_out (hwo);
for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
CaptureVoiceOut *cap = sc->cap;
struct capture_callback *cb;
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.destroy (cb->opaque);
}
}
QLIST_REMOVE(hwo, entries);
}
QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
if (hwi->enabled && hwi->pcm_ops->enable_in) {
hwi->pcm_ops->enable_in(hwi, false);
}
hwi->pcm_ops->fini_in (hwi);
QLIST_REMOVE(hwi, entries);
}
if (s->drv) {
s->drv->fini (s->drv_opaque);
s->drv = NULL;
}
if (s->dev) {
qapi_free_Audiodev(s->dev);
s->dev = NULL;
}
if (s->ts) {
timer_free(s->ts);
s->ts = NULL;
}
g_free(s);
}
void audio_cleanup(void)
{
is_cleaning_up = true;
while (!QTAILQ_EMPTY(&audio_states)) {
AudioState *s = QTAILQ_FIRST(&audio_states);
QTAILQ_REMOVE(&audio_states, s, list);
free_audio_state(s);
}
}
static const VMStateDescription vmstate_audio = {
.name = "audio",
.version_id = 1,
.minimum_version_id = 1,
.fields = (VMStateField[]) {
VMSTATE_END_OF_LIST()
}
};
static void audio_validate_opts(Audiodev *dev, Error **errp);
static AudiodevListEntry *audiodev_find(
AudiodevListHead *head, const char *drvname)
{
AudiodevListEntry *e;
QSIMPLEQ_FOREACH(e, head, next) {
if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
return e;
}
}
return NULL;
}
/*
* if we have dev, this function was called because of an -audiodev argument =>
* initialize a new state with it
* if dev == NULL => legacy implicit initialization, return the already created
* state or create a new one
*/
static AudioState *audio_init(Audiodev *dev, const char *name)
{
static bool atexit_registered;
size_t i;
int done = 0;
const char *drvname = NULL;
VMChangeStateEntry *e;
AudioState *s;
struct audio_driver *driver;
/* silence gcc warning about uninitialized variable */
AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
if (dev) {
/* -audiodev option */
legacy_config = false;
drvname = AudiodevDriver_str(dev->driver);
} else if (!QTAILQ_EMPTY(&audio_states)) {
if (!legacy_config) {
dolog("Device %s: audiodev default parameter is deprecated, please "
"specify audiodev=%s\n", name,
QTAILQ_FIRST(&audio_states)->dev->id);
}
return QTAILQ_FIRST(&audio_states);
} else {
/* legacy implicit initialization */
head = audio_handle_legacy_opts();
/*
* In case of legacy initialization, all Audiodevs in the list will have
* the same configuration (except the driver), so it does't matter which
* one we chose. We need an Audiodev to set up AudioState before we can
* init a driver. Also note that dev at this point is still in the
* list.
*/
dev = QSIMPLEQ_FIRST(&head)->dev;
audio_validate_opts(dev, &error_abort);
}
s = g_malloc0(sizeof(AudioState));
s->dev = dev;
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
if (!atexit_registered) {
atexit(audio_cleanup);
atexit_registered = true;
}
QTAILQ_INSERT_TAIL(&audio_states, s, list);
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
if (s->nb_hw_voices_out <= 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;
}
if (drvname) {
driver = audio_driver_lookup(drvname);
if (driver) {
done = !audio_driver_init(s, driver, true, dev);
} else {
dolog ("Unknown audio driver `%s'\n", drvname);
}
} else {
for (i = 0; audio_prio_list[i]; i++) {
AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
driver = audio_driver_lookup(audio_prio_list[i]);
if (e && driver) {
s->dev = dev = e->dev;
audio_validate_opts(dev, &error_abort);
done = !audio_driver_init(s, driver, false, dev);
if (done) {
e->dev = NULL;
break;
}
}
}
}
audio_free_audiodev_list(&head);
if (!done) {
driver = audio_driver_lookup("none");
done = !audio_driver_init(s, driver, false, dev);
assert(done);
dolog("warning: Using timer based audio emulation\n");
}
if (dev->timer_period <= 0) {
s->period_ticks = 1;
} else {
s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
}
e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
if (!e) {
dolog ("warning: Could not register change state handler\n"
"(Audio can continue looping even after stopping the VM)\n");
}
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
return s;
}
void audio_free_audiodev_list(AudiodevListHead *head)
{
AudiodevListEntry *e;
while ((e = QSIMPLEQ_FIRST(head))) {
QSIMPLEQ_REMOVE_HEAD(head, next);
qapi_free_Audiodev(e->dev);
g_free(e);
}
}
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
if (!card->state) {
card->state = audio_init(NULL, name);
}
card->name = g_strdup (name);
memset (&card->entries, 0, sizeof (card->entries));
QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
}
void AUD_remove_card (QEMUSoundCard *card)
{
QLIST_REMOVE (card, entries);
g_free (card->name);
}
CaptureVoiceOut *AUD_add_capture(
AudioState *s,
struct audsettings *as,
struct audio_capture_ops *ops,
void *cb_opaque
)
{
CaptureVoiceOut *cap;
struct capture_callback *cb;
if (!s) {
if (!legacy_config) {
dolog("Capturing without setting an audiodev is deprecated\n");
}
s = audio_init(NULL, NULL);
}
if (!audio_get_pdo_out(s->dev)->mixing_engine) {
dolog("Can't capture with mixeng disabled\n");
return NULL;
}
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
return NULL;
}
cb = g_malloc0(sizeof(*cb));
cb->ops = *ops;
cb->opaque = cb_opaque;
cap = audio_pcm_capture_find_specific(s, as);
if (cap) {
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
return cap;
}
else {
HWVoiceOut *hw;
CaptureVoiceOut *cap;
cap = g_malloc0(sizeof(*cap));
hw = &cap->hw;
hw->s = s;
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
/* XXX find a more elegant way */
hw->samples = 4096 * 4;
audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
if (hw->info.is_float) {
hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
} else {
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
[hw->info.is_signed]
[hw->info.swap_endianness]
[audio_bits_to_index(hw->info.bits)];
}
QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
QLIST_FOREACH(hw, &s->hw_head_out, entries) {
audio_attach_capture (hw);
}
return cap;
}
}
void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
{
struct capture_callback *cb;
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
if (cb->opaque == cb_opaque) {
cb->ops.destroy (cb_opaque);
QLIST_REMOVE (cb, entries);
g_free (cb);
if (!cap->cb_head.lh_first) {
SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
while (sw) {
SWVoiceCap *sc = (SWVoiceCap *) sw;
#ifdef DEBUG_CAPTURE
dolog ("freeing %s\n", sw->name);
#endif
sw1 = sw->entries.le_next;
if (sw->rate) {
st_rate_stop (sw->rate);
sw->rate = NULL;
}
QLIST_REMOVE (sw, entries);
QLIST_REMOVE (sc, entries);
g_free (sc);
sw = sw1;
}
QLIST_REMOVE (cap, entries);
g_free (cap->hw.mix_buf);
g_free (cap->buf);
g_free (cap);
}
return;
}
}
}
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
{
Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
audio_set_volume_out(sw, &vol);
}
void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
{
if (sw) {
HWVoiceOut *hw = sw->hw;
sw->vol.mute = vol->mute;
sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
255;
if (hw->pcm_ops->volume_out) {
hw->pcm_ops->volume_out(hw, vol);
}
}
}
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
{
Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
audio_set_volume_in(sw, &vol);
}
void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
{
if (sw) {
HWVoiceIn *hw = sw->hw;
sw->vol.mute = vol->mute;
sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
255;
if (hw->pcm_ops->volume_in) {
hw->pcm_ops->volume_in(hw, vol);
}
}
}
void audio_create_pdos(Audiodev *dev)
{
switch (dev->driver) {
#define CASE(DRIVER, driver, pdo_name) \
case AUDIODEV_DRIVER_##DRIVER: \
if (!dev->u.driver.has_in) { \
dev->u.driver.in = g_malloc0( \
sizeof(Audiodev##pdo_name##PerDirectionOptions)); \
dev->u.driver.has_in = true; \
} \
if (!dev->u.driver.has_out) { \
dev->u.driver.out = g_malloc0( \
sizeof(Audiodev##pdo_name##PerDirectionOptions)); \
dev->u.driver.has_out = true; \
} \
break
CASE(NONE, none, );
CASE(ALSA, alsa, Alsa);
CASE(COREAUDIO, coreaudio, Coreaudio);
CASE(DSOUND, dsound, );
CASE(JACK, jack, Jack);
CASE(OSS, oss, Oss);
CASE(PA, pa, Pa);
CASE(SDL, sdl, );
CASE(SPICE, spice, );
CASE(WAV, wav, );
case AUDIODEV_DRIVER__MAX:
abort();
};
}
static void audio_validate_per_direction_opts(
AudiodevPerDirectionOptions *pdo, Error **errp)
{
if (!pdo->has_mixing_engine) {
pdo->has_mixing_engine = true;
pdo->mixing_engine = true;
}
if (!pdo->has_fixed_settings) {
pdo->has_fixed_settings = true;
pdo->fixed_settings = pdo->mixing_engine;
}
if (!pdo->fixed_settings &&
(pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
error_setg(errp,
"You can't use frequency, channels or format with fixed-settings=off");
return;
}
if (!pdo->mixing_engine && pdo->fixed_settings) {
error_setg(errp, "You can't use fixed-settings without mixeng");
return;
}
if (!pdo->has_frequency) {
pdo->has_frequency = true;
pdo->frequency = 44100;
}
if (!pdo->has_channels) {
pdo->has_channels = true;
pdo->channels = 2;
}
if (!pdo->has_voices) {
pdo->has_voices = true;
pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
}
if (!pdo->has_format) {
pdo->has_format = true;
pdo->format = AUDIO_FORMAT_S16;
}
}
static void audio_validate_opts(Audiodev *dev, Error **errp)
{
Error *err = NULL;
audio_create_pdos(dev);
audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
if (err) {
error_propagate(errp, err);
return;
}
audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
if (err) {
error_propagate(errp, err);
return;
}
if (!dev->has_timer_period) {
dev->has_timer_period = true;
dev->timer_period = 10000; /* 100Hz -> 10ms */
}
}
void audio_parse_option(const char *opt)
{
AudiodevListEntry *e;
Audiodev *dev = NULL;
Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
visit_type_Audiodev(v, NULL, &dev, &error_fatal);
visit_free(v);
audio_validate_opts(dev, &error_fatal);
e = g_malloc0(sizeof(AudiodevListEntry));
e->dev = dev;
QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
}
void audio_init_audiodevs(void)
{
AudiodevListEntry *e;
QSIMPLEQ_FOREACH(e, &audiodevs, next) {
audio_init(e->dev, NULL);
}
}
audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
{
return (audsettings) {
.freq = pdo->frequency,
.nchannels = pdo->channels,
.fmt = pdo->format,
.endianness = AUDIO_HOST_ENDIANNESS,
};
}
int audioformat_bytes_per_sample(AudioFormat fmt)
{
switch (fmt) {
case AUDIO_FORMAT_U8:
case AUDIO_FORMAT_S8:
return 1;
case AUDIO_FORMAT_U16:
case AUDIO_FORMAT_S16:
return 2;
case AUDIO_FORMAT_U32:
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_F32:
return 4;
case AUDIO_FORMAT__MAX:
;
}
abort();
}
/* frames = freq * usec / 1e6 */
int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
audsettings *as, int def_usecs)
{
uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
return (as->freq * usecs + 500000) / 1000000;
}
/* samples = channels * frames = channels * freq * usec / 1e6 */
int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
audsettings *as, int def_usecs)
{
return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
}
/*
* bytes = bytes_per_sample * samples =
* bytes_per_sample * channels * freq * usec / 1e6
*/
int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
audsettings *as, int def_usecs)
{
return audio_buffer_samples(pdo, as, def_usecs) *
audioformat_bytes_per_sample(as->fmt);
}
AudioState *audio_state_by_name(const char *name)
{
AudioState *s;
QTAILQ_FOREACH(s, &audio_states, list) {
assert(s->dev);
if (strcmp(name, s->dev->id) == 0) {
return s;
}
}
return NULL;
}
const char *audio_get_id(QEMUSoundCard *card)
{
if (card->state) {
assert(card->state->dev);
return card->state->dev->id;
} else {
return "";
}
}
void audio_rate_start(RateCtl *rate)
{
memset(rate, 0, sizeof(RateCtl));
rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
}
size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
size_t bytes_avail)
{
int64_t now;
int64_t ticks;
int64_t bytes;
int64_t samples;
size_t ret;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - rate->start_ticks;
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
if (samples < 0 || samples > 65536) {
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
audio_rate_start(rate);
samples = 0;
}
ret = MIN(samples * info->bytes_per_frame, bytes_avail);
rate->bytes_sent += ret;
return ret;
}
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