blob: 8f63d41ed511345562f3fb4317a6b33787470f65 [file] [log] [blame]
/*
* QEMU ESD audio driver
*
* Copyright (c) 2008-2009 The Android Open Source Project
* 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"
#include "android/qemu-debug.h"
#define DEBUG 1
#if DEBUG
# include <stdio.h>
# define D(...) VERBOSE_PRINT(audio,__VA_ARGS__)
# define D_ACTIVE VERBOSE_CHECK(audio)
# define O(...) VERBOSE_PRINT(audioout,__VA_ARGS__)
# define I(...) VERBOSE_PRINT(audioin,__VA_ARGS__)
#else
# define D(...) ((void)0)
# define D_ACTIVE 0
# define O(...) ((void)0)
# define I(...) ((void)0)
#endif
#define STRINGIFY_(x) #x
#define STRINGIFY(x) STRINGIFY_(x)
#include <dlfcn.h>
/* link dynamically to the libesd.so */
static void* esd_lib;
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:
g_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);
g_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,
&nominal_volume);
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:
g_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);
g_free (esd->pcm_buf);
esd->pcm_buf = NULL;
}
static int qesd_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
return 0;
}
extern int esound_dynlink_init(void* lib);
/* common */
static void *qesd_audio_init (void)
{
void* result = NULL;
D("%s: entering", __FUNCTION__);
if (esd_lib == NULL) {
int fd;
esd_lib = dlopen( "libesd.so", RTLD_NOW );
if (esd_lib == NULL)
esd_lib = dlopen( "libesd.so.0", RTLD_NOW );
if (esd_lib == NULL) {
D("could not find libesd on this system");
goto Exit;
}
if (esound_dynlink_init(esd_lib) < 0)
goto Fail;
fd = esd_open_sound(conf.dac_host);
if (fd < 0) {
D("%s: could not open direct sound server connection, trying localhost",
__FUNCTION__);
fd = esd_open_sound("localhost");
if (fd < 0) {
D("%s: could not open localhost sound server connection", __FUNCTION__);
goto Fail;
}
}
D("%s: EsounD server connection succeeded", __FUNCTION__);
/* esd_close(fd); */
}
result = &conf;
goto Exit;
Fail:
D("%s: failed to open library", __FUNCTION__);
dlclose(esd_lib);
esd_lib = NULL;
Exit:
return result;
}
static void qesd_audio_fini (void *opaque)
{
(void) opaque;
if (esd_lib != NULL) {
dlclose(esd_lib);
esd_lib = NULL;
}
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)
};