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/* public domain */
#include "qemu-common.h"
#include "audio.h"
#include <pulse/simple.h>
#include <pulse/error.h>
#define AUDIO_CAP "pulseaudio"
#include "audio_int.h"
#include "audio_pt_int.h"
typedef struct {
HWVoiceOut hw;
int done;
int live;
int decr;
int rpos;
pa_simple *s;
void *pcm_buf;
struct audio_pt pt;
} PAVoiceOut;
typedef struct {
HWVoiceIn hw;
int done;
int dead;
int incr;
int wpos;
pa_simple *s;
void *pcm_buf;
struct audio_pt pt;
} PAVoiceIn;
static struct {
int samples;
char *server;
char *sink;
char *source;
} conf = {
.samples = 4096,
};
static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", pa_strerror (err));
}
static void *qpa_thread_out (void *arg)
{
PAVoiceOut *pa = arg;
HWVoiceOut *hw = &pa->hw;
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return NULL;
}
for (;;) {
int decr, to_mix, rpos;
for (;;) {
if (pa->done) {
goto exit;
}
if (pa->live > 0) {
break;
}
if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
goto exit;
}
}
decr = to_mix = audio_MIN (pa->live, conf.samples >> 2);
rpos = pa->rpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
return NULL;
}
while (to_mix) {
int error;
int chunk = audio_MIN (to_mix, hw->samples - rpos);
struct st_sample *src = hw->mix_buf + rpos;
hw->clip (pa->pcm_buf, src, chunk);
if (pa_simple_write (pa->s, pa->pcm_buf,
chunk << hw->info.shift, &error) < 0) {
qpa_logerr (error, "pa_simple_write failed\n");
return NULL;
}
rpos = (rpos + chunk) % hw->samples;
to_mix -= chunk;
}
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return NULL;
}
pa->rpos = rpos;
pa->live -= decr;
pa->decr += decr;
}
exit:
audio_pt_unlock (&pa->pt, AUDIO_FUNC);
return NULL;
}
static int qpa_run_out (HWVoiceOut *hw, int live)
{
int decr;
PAVoiceOut *pa = (PAVoiceOut *) hw;
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return 0;
}
decr = audio_MIN (live, pa->decr);
pa->decr -= decr;
pa->live = live - decr;
hw->rpos = pa->rpos;
if (pa->live > 0) {
audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
}
else {
audio_pt_unlock (&pa->pt, AUDIO_FUNC);
}
return decr;
}
static int qpa_write (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}
/* capture */
static void *qpa_thread_in (void *arg)
{
PAVoiceIn *pa = arg;
HWVoiceIn *hw = &pa->hw;
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return NULL;
}
for (;;) {
int incr, to_grab, wpos;
for (;;) {
if (pa->done) {
goto exit;
}
if (pa->dead > 0) {
break;
}
if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
goto exit;
}
}
incr = to_grab = audio_MIN (pa->dead, conf.samples >> 2);
wpos = pa->wpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
return NULL;
}
while (to_grab) {
int error;
int chunk = audio_MIN (to_grab, hw->samples - wpos);
void *buf = advance (pa->pcm_buf, wpos);
if (pa_simple_read (pa->s, buf,
chunk << hw->info.shift, &error) < 0) {
qpa_logerr (error, "pa_simple_read failed\n");
return NULL;
}
hw->conv (hw->conv_buf + wpos, buf, chunk);
wpos = (wpos + chunk) % hw->samples;
to_grab -= chunk;
}
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return NULL;
}
pa->wpos = wpos;
pa->dead -= incr;
pa->incr += incr;
}
exit:
audio_pt_unlock (&pa->pt, AUDIO_FUNC);
return NULL;
}
static int qpa_run_in (HWVoiceIn *hw)
{
int live, incr, dead;
PAVoiceIn *pa = (PAVoiceIn *) hw;
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return 0;
}
live = audio_pcm_hw_get_live_in (hw);
dead = hw->samples - live;
incr = audio_MIN (dead, pa->incr);
pa->incr -= incr;
pa->dead = dead - incr;
hw->wpos = pa->wpos;
if (pa->dead > 0) {
audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
}
else {
audio_pt_unlock (&pa->pt, AUDIO_FUNC);
}
return incr;
}
static int qpa_read (SWVoiceIn *sw, void *buf, int len)
{
return audio_pcm_sw_read (sw, buf, len);
}
static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
{
int format;
switch (afmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
format = PA_SAMPLE_U8;
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
break;
default:
dolog ("Internal logic error: Bad audio format %d\n", afmt);
format = PA_SAMPLE_U8;
break;
}
return format;
}
static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
{
switch (fmt) {
case PA_SAMPLE_U8:
return AUD_FMT_U8;
case PA_SAMPLE_S16BE:
*endianness = 1;
return AUD_FMT_S16;
case PA_SAMPLE_S16LE:
*endianness = 0;
return AUD_FMT_S16;
case PA_SAMPLE_S32BE:
*endianness = 1;
return AUD_FMT_S32;
case PA_SAMPLE_S32LE:
*endianness = 0;
return AUD_FMT_S32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
return AUD_FMT_U8;
}
}
static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
{
int error;
static pa_sample_spec ss;
static pa_buffer_attr ba;
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
/*
* qemu audio tick runs at 250 Hz (by default), so processing
* data chunks worth 4 ms of sound should be a good fit.
*/
ba.tlength = pa_usec_to_bytes (4 * 1000, &ss);
ba.minreq = pa_usec_to_bytes (2 * 1000, &ss);
ba.maxlength = -1;
ba.prebuf = -1;
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->s = pa_simple_new (
conf.server,
"qemu",
PA_STREAM_PLAYBACK,
conf.sink,
"pcm.playback",
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
&error
);
if (!pa->s) {
qpa_logerr (error, "pa_simple_new for playback failed\n");
goto fail1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
goto fail2;
}
if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) {
goto fail3;
}
return 0;
fail3:
qemu_free (pa->pcm_buf);
pa->pcm_buf = NULL;
fail2:
pa_simple_free (pa->s);
pa->s = NULL;
fail1:
return -1;
}
static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
{
int error;
static pa_sample_spec ss;
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->s = pa_simple_new (
conf.server,
"qemu",
PA_STREAM_RECORD,
conf.source,
"pcm.capture",
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
&error
);
if (!pa->s) {
qpa_logerr (error, "pa_simple_new for capture failed\n");
goto fail1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
goto fail2;
}
if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
goto fail3;
}
return 0;
fail3:
qemu_free (pa->pcm_buf);
pa->pcm_buf = NULL;
fail2:
pa_simple_free (pa->s);
pa->s = NULL;
fail1:
return -1;
}
static void qpa_fini_out (HWVoiceOut *hw)
{
void *ret;
PAVoiceOut *pa = (PAVoiceOut *) hw;
audio_pt_lock (&pa->pt, AUDIO_FUNC);
pa->done = 1;
audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
if (pa->s) {
pa_simple_free (pa->s);
pa->s = NULL;
}
audio_pt_fini (&pa->pt, AUDIO_FUNC);
qemu_free (pa->pcm_buf);
pa->pcm_buf = NULL;
}
static void qpa_fini_in (HWVoiceIn *hw)
{
void *ret;
PAVoiceIn *pa = (PAVoiceIn *) hw;
audio_pt_lock (&pa->pt, AUDIO_FUNC);
pa->done = 1;
audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
if (pa->s) {
pa_simple_free (pa->s);
pa->s = NULL;
}
audio_pt_fini (&pa->pt, AUDIO_FUNC);
qemu_free (pa->pcm_buf);
pa->pcm_buf = NULL;
}
static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
return 0;
}
static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
return 0;
}
/* common */
static void *qpa_audio_init (void)
{
return &conf;
}
static void qpa_audio_fini (void *opaque)
{
(void) opaque;
}
struct audio_option qpa_options[] = {
{
.name = "SAMPLES",
.tag = AUD_OPT_INT,
.valp = &conf.samples,
.descr = "buffer size in samples"
},
{
.name = "SERVER",
.tag = AUD_OPT_STR,
.valp = &conf.server,
.descr = "server address"
},
{
.name = "SINK",
.tag = AUD_OPT_STR,
.valp = &conf.sink,
.descr = "sink device name"
},
{
.name = "SOURCE",
.tag = AUD_OPT_STR,
.valp = &conf.source,
.descr = "source device name"
},
{ /* End of list */ }
};
static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.run_out = qpa_run_out,
.write = qpa_write,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.run_in = qpa_run_in,
.read = qpa_read,
.ctl_in = qpa_ctl_in
};
struct audio_driver pa_audio_driver = {
.name = "pa",
.descr = "http://www.pulseaudio.org/",
.options = qpa_options,
.init = qpa_audio_init,
.fini = qpa_audio_fini,
.pcm_ops = &qpa_pcm_ops,
.can_be_default = 1,
.max_voices_out = INT_MAX,
.max_voices_in = INT_MAX,
.voice_size_out = sizeof (PAVoiceOut),
.voice_size_in = sizeof (PAVoiceIn)
};
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