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42061a14358c4a26b2104c47ba1ff2adb14bbd63
qemu/audio/audio-mixeng-be.c
Marc-André Lureau 42061a1435 audio/mixeng: replace redundant pcm_info fields with AudioFormat 
The audio_pcm_info structure stored three fields (bits, is_signed,
is_float) that were always derived from the AudioFormat enum. This
redundancy meant the same information was represented twice, with no
type-level guarantee that they stayed in sync.

Replace these fields with a single AudioFormat field, and add helper
functions to extract the derived properties when needed:
- audio_format_bits()
- audio_format_is_signed()
- audio_format_is_float()

This improves type safety by making AudioFormat the single source of
truth, eliminating the possibility of inconsistent state between the
format enum and its derived boolean/integer representations.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
2026-02-23 14:38:22 +01:00

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/*
* SPDX-License-Identifier: MIT
*
* Copyright (c) 2003-2005 Vassili Karpov (malc)
*/
#include "qemu/osdep.h"
#include "qemu/audio.h"
#include "migration/vmstate.h"
#include "qemu/bswap.h"
#include "qemu/timer.h"
#include "qapi/error.h"
#include "qapi/clone-visitor.h"
#include "qapi/qobject-input-visitor.h"
#include "qapi/qapi-visit-audio.h"
#include "qapi/qapi-commands-audio.h"
#include "qobject/qdict.h"
#include "qemu/error-report.h"
#include "qemu/log.h"
#include "qemu/module.h"
#include "qemu/help_option.h"
#include "qom/object.h"
#include "system/system.h"
#include "system/replay.h"
#include "system/runstate.h"
#include "trace.h"
#define AUDIO_CAP "audio"
#include "audio_int.h"
/* #define DEBUG_OUT */
/* #define DEBUG_CAPTURE */
/* #define DEBUG_POLL */
#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
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
};
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");
}
return cond;
}
/*
* Convert audio format to mixeng_clip index. Used by audio_pcm_sw_init_ and
* audio_mixeng_backend_add_capture()
*/
static int audio_format_to_index(AudioFormat af)
{
switch (af) {
case AUDIO_FORMAT_U8:
case AUDIO_FORMAT_S8:
return 0;
case AUDIO_FORMAT_U16:
case AUDIO_FORMAT_S16:
return 1;
case AUDIO_FORMAT_U32:
case AUDIO_FORMAT_S32:
return 2;
case AUDIO_FORMAT_F32:
case AUDIO_FORMAT__MAX:
break;
}
g_assert_not_reached();
}
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 (const 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 (const 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, const struct audsettings *as)
{
return info->af == as->fmt
&& info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->swap_endianness == (as->endianness != HOST_BIG_ENDIAN);
}
void audio_pcm_init_info (struct audio_pcm_info *info, const struct audsettings *as)
{
info->af = as->fmt;
info->freq = as->freq;
info->nchannels = as->nchannels;
info->bytes_per_frame = as->nchannels * audio_format_bits(as->fmt) / 8;
info->bytes_per_second = info->freq * info->bytes_per_frame;
info->swap_endianness = (as->endianness != HOST_BIG_ENDIAN);
}
void audio_pcm_info_clear_buf(struct audio_pcm_info *info, void *buf, int len)
{
if (!len) {
return;
}
switch (info->af) {
case AUDIO_FORMAT_U8:
memset(buf, 0x80, len * info->bytes_per_frame);
break;
case AUDIO_FORMAT_U16: {
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 AUDIO_FORMAT_U32: {
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;
}
case AUDIO_FORMAT_S8:
case AUDIO_FORMAT_S16:
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_F32:
memset(buf, 0x00, len * info->bytes_per_frame);
break;
case AUDIO_FORMAT__MAX:
g_assert_not_reached();
}
}
/*
* Capture
*/
static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioMixengBackend *s,
const 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, bool 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;
bool enabled = false;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
enabled = true;
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)
{
AudioMixengBackend *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 = true;
sw->active = hw->enabled;
sw->vol = nominal_volume;
sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
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 size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
{
size_t conv = 0;
STSampleBuffer *conv_buf = &hw->conv_buf;
while (samples) {
uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->buffer + conv_buf->pos, src, proc);
conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
}
return conv;
}
/*
* Soft voice (capture)
*/
static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
size_t frames_in_max, size_t frames_out_max,
size_t *total_in, size_t *total_out)
{
HWVoiceIn *hw = sw->hw;
struct st_sample *src, *dst;
size_t live, rpos, frames_in, frames_out;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
/* resample conv_buf from rpos to end of buffer */
src = hw->conv_buf.buffer + rpos;
frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos);
dst = sw->resample_buf.buffer;
frames_out = frames_out_max;
st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
rpos += frames_in;
*total_in = frames_in;
*total_out = frames_out;
/* resample conv_buf from start of buffer if there are input frames left */
if (frames_in_max - frames_in && rpos == hw->conv_buf.size) {
src = hw->conv_buf.buffer;
frames_in = frames_in_max - frames_in;
dst += frames_out;
frames_out = frames_out_max - frames_out;
st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
*total_in += frames_in;
*total_out += frames_out;
}
}
static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
{
HWVoiceIn *hw = sw->hw;
size_t live, frames_out_max, total_in, total_out;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (!live) {
return 0;
}
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;
}
frames_out_max = MIN(buf_len / sw->info.bytes_per_frame,
sw->resample_buf.size);
audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out);
if (!AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s)->volume_in) {
mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol);
}
sw->clip(buf, sw->resample_buf.buffer, total_out);
sw->total_hw_samples_acquired += total_in;
return total_out * 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;
}
static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
return (k->buffer_get_free ? k->buffer_get_free(hw) : INT_MAX) /
hw->info.bytes_per_frame;
}
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.buffer + 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 (playback)
*/
static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
size_t frames_in_max, size_t frames_out_max,
size_t *total_in, size_t *total_out)
{
HWVoiceOut *hw = sw->hw;
struct st_sample *src, *dst;
size_t live, wpos, frames_in, frames_out;
live = sw->total_hw_samples_mixed;
wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size;
/* write to mix_buf from wpos to end of buffer */
src = sw->resample_buf.buffer;
frames_in = frames_in_max;
dst = hw->mix_buf.buffer + wpos;
frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos);
st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
wpos += frames_out;
*total_in = frames_in;
*total_out = frames_out;
/* write to mix_buf from start of buffer if there are input frames left */
if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) {
src += frames_in;
frames_in = frames_in_max - frames_in;
dst = hw->mix_buf.buffer;
frames_out = frames_out_max - frames_out;
st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
*total_in += frames_in;
*total_out += frames_out;
}
}
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
{
HWVoiceOut *hw = sw->hw;
size_t live, dead, hw_free, sw_max, fe_max;
size_t frames_in_max, frames_out_max, total_in, total_out;
live = sw->total_hw_samples_mixed;
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;
}
if (live == hw->mix_buf.size) {
#ifdef DEBUG_OUT
dolog ("%s is full %zu\n", sw->name, live);
#endif
return 0;
}
dead = hw->mix_buf.size - live;
hw_free = audio_pcm_hw_get_free(hw);
hw_free = hw_free > live ? hw_free - live : 0;
frames_out_max = MIN(dead, hw_free);
sw_max = st_rate_frames_in(sw->rate, frames_out_max);
fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos,
sw->resample_buf.size);
frames_in_max = MIN(sw_max, fe_max);
if (!frames_in_max) {
return 0;
}
if (frames_in_max > sw->resample_buf.pos) {
sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos,
buf, frames_in_max - sw->resample_buf.pos);
if (!AUDIO_MIXENG_BACKEND_GET_CLASS(sw->hw->s)->volume_out) {
mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos,
frames_in_max - sw->resample_buf.pos, &sw->vol);
}
}
audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
&total_in, &total_out);
sw->total_hw_samples_mixed += total_out;
sw->empty = sw->total_hw_samples_mixed == 0;
/*
* Upsampling may leave one audio frame in the resample buffer. Decrement
* total_in by one if there was a leftover frame from the previous resample
* pass in the resample buffer. Increment total_in by one if the current
* resample pass left one frame in the resample buffer.
*/
if (frames_in_max - total_in == 1) {
/* copy one leftover audio frame to the beginning of the buffer */
*sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in);
total_in += 1 - sw->resample_buf.pos;
sw->resample_buf.pos = 1;
} else if (total_in >= sw->resample_buf.pos) {
total_in -= sw->resample_buf.pos;
sw->resample_buf.pos = 0;
}
#ifdef DEBUG_OUT
dolog (
"%s: write size %zu written %zu total mixed %zu\n",
SW_NAME(sw),
buf_len / sw->info.bytes_per_frame,
total_in,
sw->total_hw_samples_mixed
);
#endif
return total_in * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
{
dolog("%s: %s, freq %d, nchan %d\n",
cap, AudioFormat_str(info->af), info->freq,
info->nchannels);
}
#endif
#define DAC
#include "audio_template.h"
#undef DAC
#include "audio_template.h"
/*
* Timer
*/
static int audio_is_timer_needed(AudioMixengBackend *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(AudioMixengBackend *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;
AudioMixengBackend *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
*/
static size_t audio_mixeng_backend_write(AudioBackend *be, 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 AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s)->write(hw, buf, size);
}
}
static size_t audio_mixeng_backend_read(AudioBackend *be,
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 AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s)->read(hw, buf, size);
}
}
static int audio_mixeng_backend_get_buffer_size_out(AudioBackend *be, SWVoiceOut *sw)
{
if (!sw) {
return 0;
}
if (audio_get_pdo_out(sw->s->dev)->mixing_engine) {
return sw->resample_buf.size * sw->info.bytes_per_frame;
}
return sw->hw->samples * sw->hw->info.bytes_per_frame;
}
static void audio_mixeng_backend_set_active_out(AudioBackend *be, SWVoiceOut *sw,
bool on)
{
HWVoiceOut *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioMixengBackend *s = sw->s;
SWVoiceOut *temp_sw;
SWVoiceCap *sc;
if (on) {
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(s);
hw->pending_disable = 0;
if (!hw->enabled) {
hw->enabled = true;
if (runstate_is_running()) {
if (k->enable_out) {
k->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;
}
}
static void audio_mixeng_backend_set_active_in(AudioBackend *be, SWVoiceIn *sw, bool on)
{
HWVoiceIn *hw;
if (!sw) {
return;
}
hw = sw->hw;
if (sw->active != on) {
AudioMixengBackend *s = sw->s;
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(s);
SWVoiceIn *temp_sw;
if (on) {
if (!hw->enabled) {
hw->enabled = true;
if (runstate_is_running()) {
if (k->enable_in) {
k->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 = false;
if (k->enable_in) {
k->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 %zu frontend frames %u\n",
SW_NAME (sw),
live, st_rate_frames_out(sw->rate, live)
);
return live;
}
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 %zu dead %zu frontend frames %u\n",
SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead));
#endif
return dead;
}
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;
size_t rpos2 = rpos;
n = samples;
while (n) {
size_t till_end_of_hw = hw->mix_buf.size - rpos2;
size_t to_read = MIN(till_end_of_hw, n);
size_t live, frames_in, frames_out;
sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
sw->resample_buf.size = to_read;
live = sw->total_hw_samples_mixed;
audio_pcm_sw_resample_out(sw,
to_read, sw->hw->mix_buf.size - live,
&frames_in, &frames_out);
sw->total_hw_samples_mixed += frames_out;
sw->empty = sw->total_hw_samples_mixed == 0;
if (to_read - frames_in) {
dolog("Could not mix %zu frames into a capture "
"buffer, mixed %zu\n",
to_read, frames_in);
break;
}
n -= to_read;
rpos2 = (rpos2 + to_read) % hw->mix_buf.size;
}
}
}
n = MIN(samples, hw->mix_buf.size - rpos);
mixeng_clear(hw->mix_buf.buffer + rpos, n);
mixeng_clear(hw->mix_buf.buffer, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
size_t clipped = 0;
while (live) {
size_t size = live * hw->info.bytes_per_frame;
size_t decr, proc;
void *buf = k->get_buffer_out(hw, &size);
if (size == 0) {
break;
}
decr = MIN(size / hw->info.bytes_per_frame, live);
if (buf) {
audio_pcm_hw_clip_out(hw, buf, decr);
}
proc = k->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 (k->run_buffer_out) {
k->run_buffer_out(hw);
}
return clipped;
}
static void audio_run_out(AudioMixengBackend *s)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(s);
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
size_t played, live, prev_rpos;
size_t hw_free = audio_pcm_hw_get_free(hw);
int nb_live;
if (!audio_get_pdo_out(s->dev)->mixing_engine) {
/* there is exactly 1 sw for each hw with no mixeng */
sw = hw->sw_head.lh_first;
if (hw->pending_disable) {
hw->enabled = false;
hw->pending_disable = false;
if (k->enable_out) {
k->enable_out(hw, false);
}
}
if (sw->active) {
sw->callback.fn(sw->callback.opaque,
hw_free * sw->info.bytes_per_frame);
}
if (k->run_buffer_out) {
k->run_buffer_out(hw);
}
continue;
}
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
size_t sw_free = audio_get_free(sw);
size_t free;
if (hw_free > sw->total_hw_samples_mixed) {
free = st_rate_frames_in(sw->rate,
MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
} else {
free = 0;
}
if (free > sw->resample_buf.pos) {
free = MIN(free, sw->resample_buf.size)
- sw->resample_buf.pos;
sw->callback.fn(sw->callback.opaque,
free * sw->info.bytes_per_frame);
}
}
}
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 = false;
hw->pending_disable = false;
if (k->enable_out) {
k->enable_out(hw, false);
}
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = false;
audio_recalc_and_notify_capture (sc->cap);
}
continue;
}
if (!live) {
if (k->run_buffer_out) {
k->run_buffer_out(hw);
}
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) {
audio_capture_mix_and_clear (hw, prev_rpos, played);
}
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 = true;
}
}
}
}
static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
size_t conv = 0;
if (k->run_buffer_in) {
k->run_buffer_in(hw);
}
while (samples) {
size_t proc;
size_t size = samples * hw->info.bytes_per_frame;
void *buf = k->get_buffer_in(hw, &size);
assert(size % hw->info.bytes_per_frame == 0);
if (size == 0) {
break;
}
proc = audio_pcm_hw_conv_in(hw, buf, size / hw->info.bytes_per_frame);
samples -= proc;
conv += proc;
k->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
}
return conv;
}
static void audio_run_in(AudioMixengBackend *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;
int pos;
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_start(&captured);
assert(captured <= hw->conv_buf.size);
if (replay_mode == REPLAY_MODE_PLAY) {
hw->conv_buf.pos = (hw->conv_buf.pos + captured) % hw->conv_buf.size;
}
for (pos = (hw->conv_buf.pos - captured + hw->conv_buf.size) % hw->conv_buf.size;
pos != hw->conv_buf.pos;
pos = (pos + 1) % hw->conv_buf.size) {
uint64_t left, right;
if (replay_mode == REPLAY_MODE_RECORD) {
audio_sample_to_uint64(hw->conv_buf.buffer, pos, &left, &right);
}
replay_audio_in_sample_lr(&left, &right);
if (replay_mode == REPLAY_MODE_PLAY) {
audio_sample_from_uint64(hw->conv_buf.buffer, pos, left, right);
}
}
replay_audio_in_finish();
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
sw->total_hw_samples_acquired -= min;
if (sw->active) {
size_t sw_avail = audio_get_avail(sw);
size_t avail;
avail = st_rate_frames_out(sw->rate, sw_avail);
if (avail > 0) {
avail = MIN(avail, sw->resample_buf.size);
sw->callback.fn(sw->callback.opaque,
avail * sw->info.bytes_per_frame);
}
}
}
}
}
static void audio_run_capture(AudioMixengBackend *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.buffer + 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(AudioMixengBackend *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_run_buffer_in(HWVoiceIn *hw)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
if (unlikely(!hw->buf_emul)) {
hw->size_emul = hw->samples * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(hw->size_emul);
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 = k->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;
}
}
}
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
{
size_t start;
start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
assert(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;
}
size_t audio_generic_buffer_get_free(HWVoiceOut *hw)
{
if (hw->buf_emul) {
return hw->size_emul - hw->pending_emul;
} else {
return hw->samples * hw->info.bytes_per_frame;
}
}
void audio_generic_run_buffer_out(HWVoiceOut *hw)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
while (hw->pending_emul) {
size_t write_len, written, start;
start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
assert(start < hw->size_emul);
write_len = MIN(hw->pending_emul, hw->size_emul - start);
written = k->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)) {
hw->size_emul = hw->samples * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(hw->size_emul);
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)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
size_t total = 0;
if (k->buffer_get_free) {
size_t free = k->buffer_get_free(hw);
size = MIN(size, free);
}
while (total < size) {
size_t dst_size = size - total;
size_t copy_size, proc;
void *dst = k->get_buffer_out(hw, &dst_size);
if (dst_size == 0) {
break;
}
copy_size = MIN(size - total, dst_size);
if (dst) {
memcpy(dst, (char *)buf + total, copy_size);
}
proc = k->put_buffer_out(hw, dst, copy_size);
total += proc;
if (proc == 0 || proc < copy_size) {
break;
}
}
return total;
}
size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
{
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
size_t total = 0;
if (k->run_buffer_in) {
k->run_buffer_in(hw);
}
while (total < size) {
size_t src_size = size - total;
void *src = k->get_buffer_in(hw, &src_size);
if (src_size == 0) {
break;
}
memcpy((char *)buf + total, src, src_size);
k->put_buffer_in(hw, src, src_size);
total += src_size;
}
return total;
}
static bool audio_mixeng_backend_realize(AudioBackend *abe,
Audiodev *dev, Error **errp)
{
AudioMixengBackend *be = AUDIO_MIXENG_BACKEND(abe);
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(be);
be->dev = dev;
if (!k->get_buffer_in) {
k->get_buffer_in = audio_generic_get_buffer_in;
k->put_buffer_in = audio_generic_put_buffer_in;
}
if (!k->get_buffer_out) {
k->get_buffer_out = audio_generic_get_buffer_out;
k->put_buffer_out = audio_generic_put_buffer_out;
}
audio_init_nb_voices_out(be, k, 1);
audio_init_nb_voices_in(be, k, 0);
if (be->dev->timer_period <= 0) {
be->period_ticks = 1;
} else {
be->period_ticks = be->dev->timer_period * (int64_t)SCALE_US;
}
return true;
}
static void audio_vm_change_state_handler (void *opaque, bool running,
RunState state)
{
AudioMixengBackend *s = opaque;
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(s);
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
if (k->enable_out) {
k->enable_out(hwo, running);
}
}
while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
if (k->enable_in) {
k->enable_in(hwi, running);
}
}
audio_reset_timer (s);
}
static const VMStateDescription vmstate_audio;
static const char *audio_mixeng_backend_get_id(AudioBackend *be)
{
return AUDIO_MIXENG_BACKEND(be)->dev->id;
}
static CaptureVoiceOut *audio_mixeng_backend_add_capture(
AudioBackend *be,
const struct audsettings *as,
const struct audio_capture_ops *ops,
void *cb_opaque);
static void audio_mixeng_backend_del_capture(
AudioBackend *be,
CaptureVoiceOut *cap,
void *cb_opaque);
static void audio_mixeng_backend_set_volume_out(AudioBackend *be, SWVoiceOut *sw,
Volume *vol);
static void audio_mixeng_backend_set_volume_in(AudioBackend *be, SWVoiceIn *sw,
Volume *vol);
static void audio_mixeng_backend_class_init(ObjectClass *klass, const void *data)
{
AudioBackendClass *be = AUDIO_BACKEND_CLASS(klass);
be->realize = audio_mixeng_backend_realize;
be->get_id = audio_mixeng_backend_get_id;
be->open_in = audio_mixeng_backend_open_in;
be->open_out = audio_mixeng_backend_open_out;
be->close_in = audio_mixeng_backend_close_in;
be->close_out = audio_mixeng_backend_close_out;
be->is_active_out = audio_mixeng_backend_is_active_out;
be->is_active_in = audio_mixeng_backend_is_active_in;
be->set_active_out = audio_mixeng_backend_set_active_out;
be->set_active_in = audio_mixeng_backend_set_active_in;
be->set_volume_out = audio_mixeng_backend_set_volume_out;
be->set_volume_in = audio_mixeng_backend_set_volume_in;
be->read = audio_mixeng_backend_read;
be->write = audio_mixeng_backend_write;
be->get_buffer_size_out = audio_mixeng_backend_get_buffer_size_out;
be->add_capture = audio_mixeng_backend_add_capture;
be->del_capture = audio_mixeng_backend_del_capture;
}
static void audio_mixeng_backend_init(Object *obj)
{
AudioMixengBackend *s = AUDIO_MIXENG_BACKEND(obj);
QLIST_INIT(&s->hw_head_out);
QLIST_INIT(&s->hw_head_in);
QLIST_INIT(&s->cap_head);
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
s->vmse = qemu_add_vm_change_state_handler(audio_vm_change_state_handler, s);
assert(s->vmse != NULL);
vmstate_register_any(NULL, &vmstate_audio, s);
}
static void audio_mixeng_backend_finalize(Object *obj)
{
AudioMixengBackend *s = AUDIO_MIXENG_BACKEND(obj);
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(s);
HWVoiceOut *hwo, *hwon;
HWVoiceIn *hwi, *hwin;
QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
if (hwo->enabled && k->enable_out) {
k->enable_out(hwo, false);
}
k->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 && k->enable_in) {
k->enable_in(hwi, false);
}
k->fini_in(hwi);
QLIST_REMOVE(hwi, entries);
}
if (s->dev) {
qapi_free_Audiodev(s->dev);
s->dev = NULL;
}
if (s->ts) {
timer_free(s->ts);
s->ts = NULL;
}
if (s->vmse) {
qemu_del_vm_change_state_handler(s->vmse);
s->vmse = NULL;
}
vmstate_unregister(NULL, &vmstate_audio, s);
}
static bool vmstate_audio_needed(void *opaque)
{
/*
* Never needed, this vmstate only exists in case
* an old qemu sends it to us.
*/
return false;
}
static const VMStateDescription vmstate_audio = {
.name = "audio",
.version_id = 1,
.minimum_version_id = 1,
.needed = vmstate_audio_needed,
.fields = (const VMStateField[]) {
VMSTATE_END_OF_LIST()
}
};
static CaptureVoiceOut *audio_mixeng_backend_add_capture(
AudioBackend *be,
const struct audsettings *as,
const struct audio_capture_ops *ops,
void *cb_opaque)
{
AudioMixengBackend *s = AUDIO_MIXENG_BACKEND(be);
CaptureVoiceOut *cap;
struct capture_callback *cb;
if (!s) {
error_report("Capturing without setting an audiodev is not supported");
abort();
}
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);
} else {
HWVoiceOut *hw;
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 (audio_format_is_float(hw->info.af)) {
hw->clip = mixeng_clip_float[hw->info.nchannels == 2]
[hw->info.swap_endianness];
} else {
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
[audio_format_is_signed(hw->info.af)]
[hw->info.swap_endianness]
[audio_format_to_index(hw->info.af)];
}
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;
}
static void audio_mixeng_backend_del_capture(
AudioBackend *be,
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.buffer);
g_free (cap->buf);
g_free (cap);
}
return;
}
}
}
static void audio_mixeng_backend_set_volume_out(AudioBackend *be, SWVoiceOut *sw,
Volume *vol)
{
if (sw) {
HWVoiceOut *hw = sw->hw;
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
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 (k->volume_out) {
k->volume_out(hw, vol);
}
}
}
static void audio_mixeng_backend_set_volume_in(AudioBackend *be, SWVoiceIn *sw,
Volume *vol)
{
if (sw) {
HWVoiceIn *hw = sw->hw;
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_GET_CLASS(hw->s);
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 (k->volume_in) {
k->volume_in(hw, vol);
}
}
}
audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
{
return (audsettings) {
.freq = pdo->frequency,
.nchannels = pdo->channels,
.fmt = pdo->format,
.endianness = HOST_BIG_ENDIAN,
};
}
/* 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) * audio_format_bits(as->fmt) / 8;
}
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_peek_bytes(RateCtl *rate, struct audio_pcm_info *info)
{
int64_t now;
int64_t ticks;
int64_t bytes;
int64_t frames;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - rate->start_ticks;
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
frames = (bytes - rate->bytes_sent) / info->bytes_per_frame;
rate->peeked_frames = frames;
return frames < 0 ? 0 : frames * info->bytes_per_frame;
}
void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used)
{
if (rate->peeked_frames < 0 || rate->peeked_frames > 65536) {
AUD_log(NULL, "Resetting rate control (%" PRId64 " frames)\n",
rate->peeked_frames);
audio_rate_start(rate);
}
rate->bytes_sent += bytes_used;
}
size_t audio_rate_get_bytes(RateCtl *rate, struct audio_pcm_info *info,
size_t bytes_avail)
{
size_t bytes;
bytes = audio_rate_peek_bytes(rate, info);
bytes = MIN(bytes, bytes_avail);
audio_rate_add_bytes(rate, bytes);
return bytes;
}
static const TypeInfo audio_types[] = {
{
.name = TYPE_AUDIO_MIXENG_BACKEND,
.parent = TYPE_AUDIO_BACKEND,
.instance_size = sizeof(AudioMixengBackend),
.instance_init = audio_mixeng_backend_init,
.instance_finalize = audio_mixeng_backend_finalize,
.class_size = sizeof(AudioMixengBackendClass),
.class_init = audio_mixeng_backend_class_init,
},
};
DEFINE_TYPES(audio_types)
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