qemu/audio/esdaudio.c
Michael Walle 00e076795f audio: split sample conversion and volume mixing
Refactor the volume mixing, so it can be reused for capturing devices.
Additionally, it removes superfluous multiplications with the nominal
volume within the hardware voice code path.

Signed-off-by: Michael Walle <michael@walle.cc>
Signed-off-by: malc <av1474@comtv.ru>
2011-01-12 18:36:22 +03:00

558 lines
13 KiB
C

/*
* QEMU ESD audio driver
*
* Copyright (c) 2006 Frederick Reeve (brushed up by 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 <esd.h>
#include "qemu-common.h"
#include "audio.h"
#define AUDIO_CAP "esd"
#include "audio_int.h"
#include "audio_pt_int.h"
typedef struct {
HWVoiceOut hw;
int done;
int live;
int decr;
int rpos;
void *pcm_buf;
int fd;
struct audio_pt pt;
} ESDVoiceOut;
typedef struct {
HWVoiceIn hw;
int done;
int dead;
int incr;
int wpos;
void *pcm_buf;
int fd;
struct audio_pt pt;
} ESDVoiceIn;
static struct {
int samples;
int divisor;
char *dac_host;
char *adc_host;
} conf = {
.samples = 1024,
.divisor = 2,
};
static void GCC_FMT_ATTR (2, 3) qesd_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", strerror (err));
}
/* playback */
static void *qesd_thread_out (void *arg)
{
ESDVoiceOut *esd = arg;
HWVoiceOut *hw = &esd->hw;
int threshold;
threshold = conf.divisor ? hw->samples / conf.divisor : 0;
if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
return NULL;
}
for (;;) {
int decr, to_mix, rpos;
for (;;) {
if (esd->done) {
goto exit;
}
if (esd->live > threshold) {
break;
}
if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) {
goto exit;
}
}
decr = to_mix = esd->live;
rpos = hw->rpos;
if (audio_pt_unlock (&esd->pt, AUDIO_FUNC)) {
return NULL;
}
while (to_mix) {
ssize_t written;
int chunk = audio_MIN (to_mix, hw->samples - rpos);
struct st_sample *src = hw->mix_buf + rpos;
hw->clip (esd->pcm_buf, src, chunk);
again:
written = write (esd->fd, esd->pcm_buf, chunk << hw->info.shift);
if (written == -1) {
if (errno == EINTR || errno == EAGAIN) {
goto again;
}
qesd_logerr (errno, "write failed\n");
return NULL;
}
if (written != chunk << hw->info.shift) {
int wsamples = written >> hw->info.shift;
int wbytes = wsamples << hw->info.shift;
if (wbytes != written) {
dolog ("warning: Misaligned write %d (requested %zd), "
"alignment %d\n",
wbytes, written, hw->info.align + 1);
}
to_mix -= wsamples;
rpos = (rpos + wsamples) % hw->samples;
break;
}
rpos = (rpos + chunk) % hw->samples;
to_mix -= chunk;
}
if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
return NULL;
}
esd->rpos = rpos;
esd->live -= decr;
esd->decr += decr;
}
exit:
audio_pt_unlock (&esd->pt, AUDIO_FUNC);
return NULL;
}
static int qesd_run_out (HWVoiceOut *hw, int live)
{
int decr;
ESDVoiceOut *esd = (ESDVoiceOut *) hw;
if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
return 0;
}
decr = audio_MIN (live, esd->decr);
esd->decr -= decr;
esd->live = live - decr;
hw->rpos = esd->rpos;
if (esd->live > 0) {
audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
}
else {
audio_pt_unlock (&esd->pt, AUDIO_FUNC);
}
return decr;
}
static int qesd_write (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}
static int qesd_init_out (HWVoiceOut *hw, struct audsettings *as)
{
ESDVoiceOut *esd = (ESDVoiceOut *) hw;
struct audsettings obt_as = *as;
int esdfmt = ESD_STREAM | ESD_PLAY;
esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO;
switch (as->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
esdfmt |= ESD_BITS8;
obt_as.fmt = AUD_FMT_U8;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
dolog ("Will use 16 instead of 32 bit samples\n");
case AUD_FMT_S16:
case AUD_FMT_U16:
deffmt:
esdfmt |= ESD_BITS16;
obt_as.fmt = AUD_FMT_S16;
break;
default:
dolog ("Internal logic error: Bad audio format %d\n", as->fmt);
goto deffmt;
}
obt_as.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!esd->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
return -1;
}
esd->fd = esd_play_stream (esdfmt, as->freq, conf.dac_host, NULL);
if (esd->fd < 0) {
qesd_logerr (errno, "esd_play_stream failed\n");
goto fail1;
}
if (audio_pt_init (&esd->pt, qesd_thread_out, esd, AUDIO_CAP, AUDIO_FUNC)) {
goto fail2;
}
return 0;
fail2:
if (close (esd->fd)) {
qesd_logerr (errno, "%s: close on esd socket(%d) failed\n",
AUDIO_FUNC, esd->fd);
}
esd->fd = -1;
fail1:
qemu_free (esd->pcm_buf);
esd->pcm_buf = NULL;
return -1;
}
static void qesd_fini_out (HWVoiceOut *hw)
{
void *ret;
ESDVoiceOut *esd = (ESDVoiceOut *) hw;
audio_pt_lock (&esd->pt, AUDIO_FUNC);
esd->done = 1;
audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
audio_pt_join (&esd->pt, &ret, AUDIO_FUNC);
if (esd->fd >= 0) {
if (close (esd->fd)) {
qesd_logerr (errno, "failed to close esd socket\n");
}
esd->fd = -1;
}
audio_pt_fini (&esd->pt, AUDIO_FUNC);
qemu_free (esd->pcm_buf);
esd->pcm_buf = NULL;
}
static int qesd_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
return 0;
}
/* capture */
static void *qesd_thread_in (void *arg)
{
ESDVoiceIn *esd = arg;
HWVoiceIn *hw = &esd->hw;
int threshold;
threshold = conf.divisor ? hw->samples / conf.divisor : 0;
if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
return NULL;
}
for (;;) {
int incr, to_grab, wpos;
for (;;) {
if (esd->done) {
goto exit;
}
if (esd->dead > threshold) {
break;
}
if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) {
goto exit;
}
}
incr = to_grab = esd->dead;
wpos = hw->wpos;
if (audio_pt_unlock (&esd->pt, AUDIO_FUNC)) {
return NULL;
}
while (to_grab) {
ssize_t nread;
int chunk = audio_MIN (to_grab, hw->samples - wpos);
void *buf = advance (esd->pcm_buf, wpos);
again:
nread = read (esd->fd, buf, chunk << hw->info.shift);
if (nread == -1) {
if (errno == EINTR || errno == EAGAIN) {
goto again;
}
qesd_logerr (errno, "read failed\n");
return NULL;
}
if (nread != chunk << hw->info.shift) {
int rsamples = nread >> hw->info.shift;
int rbytes = rsamples << hw->info.shift;
if (rbytes != nread) {
dolog ("warning: Misaligned write %d (requested %zd), "
"alignment %d\n",
rbytes, nread, hw->info.align + 1);
}
to_grab -= rsamples;
wpos = (wpos + rsamples) % hw->samples;
break;
}
hw->conv (hw->conv_buf + wpos, buf, nread >> hw->info.shift);
wpos = (wpos + chunk) % hw->samples;
to_grab -= chunk;
}
if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
return NULL;
}
esd->wpos = wpos;
esd->dead -= incr;
esd->incr += incr;
}
exit:
audio_pt_unlock (&esd->pt, AUDIO_FUNC);
return NULL;
}
static int qesd_run_in (HWVoiceIn *hw)
{
int live, incr, dead;
ESDVoiceIn *esd = (ESDVoiceIn *) hw;
if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
return 0;
}
live = audio_pcm_hw_get_live_in (hw);
dead = hw->samples - live;
incr = audio_MIN (dead, esd->incr);
esd->incr -= incr;
esd->dead = dead - incr;
hw->wpos = esd->wpos;
if (esd->dead > 0) {
audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
}
else {
audio_pt_unlock (&esd->pt, AUDIO_FUNC);
}
return incr;
}
static int qesd_read (SWVoiceIn *sw, void *buf, int len)
{
return audio_pcm_sw_read (sw, buf, len);
}
static int qesd_init_in (HWVoiceIn *hw, struct audsettings *as)
{
ESDVoiceIn *esd = (ESDVoiceIn *) hw;
struct audsettings obt_as = *as;
int esdfmt = ESD_STREAM | ESD_RECORD;
esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO;
switch (as->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
esdfmt |= ESD_BITS8;
obt_as.fmt = AUD_FMT_U8;
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
esdfmt |= ESD_BITS16;
obt_as.fmt = AUD_FMT_S16;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
dolog ("Will use 16 instead of 32 bit samples\n");
esdfmt |= ESD_BITS16;
obt_as.fmt = AUD_FMT_S16;
break;
}
obt_as.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!esd->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
return -1;
}
esd->fd = esd_record_stream (esdfmt, as->freq, conf.adc_host, NULL);
if (esd->fd < 0) {
qesd_logerr (errno, "esd_record_stream failed\n");
goto fail1;
}
if (audio_pt_init (&esd->pt, qesd_thread_in, esd, AUDIO_CAP, AUDIO_FUNC)) {
goto fail2;
}
return 0;
fail2:
if (close (esd->fd)) {
qesd_logerr (errno, "%s: close on esd socket(%d) failed\n",
AUDIO_FUNC, esd->fd);
}
esd->fd = -1;
fail1:
qemu_free (esd->pcm_buf);
esd->pcm_buf = NULL;
return -1;
}
static void qesd_fini_in (HWVoiceIn *hw)
{
void *ret;
ESDVoiceIn *esd = (ESDVoiceIn *) hw;
audio_pt_lock (&esd->pt, AUDIO_FUNC);
esd->done = 1;
audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
audio_pt_join (&esd->pt, &ret, AUDIO_FUNC);
if (esd->fd >= 0) {
if (close (esd->fd)) {
qesd_logerr (errno, "failed to close esd socket\n");
}
esd->fd = -1;
}
audio_pt_fini (&esd->pt, AUDIO_FUNC);
qemu_free (esd->pcm_buf);
esd->pcm_buf = NULL;
}
static int qesd_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
return 0;
}
/* common */
static void *qesd_audio_init (void)
{
return &conf;
}
static void qesd_audio_fini (void *opaque)
{
(void) opaque;
ldebug ("esd_fini");
}
struct audio_option qesd_options[] = {
{
.name = "SAMPLES",
.tag = AUD_OPT_INT,
.valp = &conf.samples,
.descr = "buffer size in samples"
},
{
.name = "DIVISOR",
.tag = AUD_OPT_INT,
.valp = &conf.divisor,
.descr = "threshold divisor"
},
{
.name = "DAC_HOST",
.tag = AUD_OPT_STR,
.valp = &conf.dac_host,
.descr = "playback host"
},
{
.name = "ADC_HOST",
.tag = AUD_OPT_STR,
.valp = &conf.adc_host,
.descr = "capture host"
},
{ /* End of list */ }
};
static struct audio_pcm_ops qesd_pcm_ops = {
.init_out = qesd_init_out,
.fini_out = qesd_fini_out,
.run_out = qesd_run_out,
.write = qesd_write,
.ctl_out = qesd_ctl_out,
.init_in = qesd_init_in,
.fini_in = qesd_fini_in,
.run_in = qesd_run_in,
.read = qesd_read,
.ctl_in = qesd_ctl_in,
};
struct audio_driver esd_audio_driver = {
.name = "esd",
.descr = "http://en.wikipedia.org/wiki/Esound",
.options = qesd_options,
.init = qesd_audio_init,
.fini = qesd_audio_fini,
.pcm_ops = &qesd_pcm_ops,
.can_be_default = 0,
.max_voices_out = INT_MAX,
.max_voices_in = INT_MAX,
.voice_size_out = sizeof (ESDVoiceOut),
.voice_size_in = sizeof (ESDVoiceIn)
};