* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
+#include "qemu/osdep.h"
#include <alsa/asoundlib.h>
#include "qemu-common.h"
-#include "qemu-char.h"
+#include "qemu/main-loop.h"
#include "audio.h"
+#include "trace.h"
-#if QEMU_GNUC_PREREQ(4, 3)
#pragma GCC diagnostic ignored "-Waddress"
-#endif
#define AUDIO_CAP "alsa"
#include "audio_int.h"
+typedef struct ALSAConf {
+ int size_in_usec_in;
+ int size_in_usec_out;
+ const char *pcm_name_in;
+ const char *pcm_name_out;
+ unsigned int buffer_size_in;
+ unsigned int period_size_in;
+ unsigned int buffer_size_out;
+ unsigned int period_size_out;
+ unsigned int threshold;
+
+ int buffer_size_in_overridden;
+ int period_size_in_overridden;
+
+ int buffer_size_out_overridden;
+ int period_size_out_overridden;
+} ALSAConf;
+
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
+ ALSAConf *conf;
int count;
+ int mask;
};
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
+ int wpos;
+ int pending;
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
struct pollhlp pollhlp;
} ALSAVoiceIn;
-static struct {
- int size_in_usec_in;
- int size_in_usec_out;
- const char *pcm_name_in;
- const char *pcm_name_out;
- unsigned int buffer_size_in;
- unsigned int period_size_in;
- unsigned int buffer_size_out;
- unsigned int period_size_out;
- unsigned int threshold;
-
- int buffer_size_in_overridden;
- int period_size_in_overridden;
-
- int buffer_size_out_overridden;
- int period_size_out_overridden;
- int verbose;
-} conf = {
- .buffer_size_out = 1024,
- .pcm_name_out = "default",
- .pcm_name_in = "default",
-};
-
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
for (i = 0; i < hlp->count; ++i) {
qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
}
- qemu_free (pfds);
+ g_free (pfds);
}
hlp->pfds = NULL;
hlp->count = 0;
return;
}
- if (!(revents & POLLOUT)) {
- if (conf.verbose) {
- dolog ("revents = %d\n", revents);
- }
+ if (!(revents & hlp->mask)) {
+ trace_alsa_revents(revents);
return;
}
state = snd_pcm_state (hlp->handle);
switch (state) {
+ case SND_PCM_STATE_SETUP:
+ alsa_recover (hlp->handle);
+ break;
+
case SND_PCM_STATE_XRUN:
alsa_recover (hlp->handle);
break;
}
}
-static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp)
+static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
{
int i, count, err;
struct pollfd *pfds;
if (err < 0) {
alsa_logerr (err, "Could not initialize poll mode\n"
"Could not obtain poll descriptors\n");
- qemu_free (pfds);
+ g_free (pfds);
return -1;
}
for (i = 0; i < count; ++i) {
if (pfds[i].events & POLLIN) {
- err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler,
- NULL, hlp);
+ qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
}
if (pfds[i].events & POLLOUT) {
- if (conf.verbose) {
- dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
- }
- err = qemu_set_fd_handler (pfds[i].fd, NULL,
- alsa_poll_handler, hlp);
+ trace_alsa_pollout(i, pfds[i].fd);
+ qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
}
- if (conf.verbose) {
- dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
- pfds[i].events, i, pfds[i].fd, err);
- }
-
- if (err) {
- dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
- pfds[i].events, i, pfds[i].fd, err);
+ trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
- while (i--) {
- qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
- }
- qemu_free (pfds);
- return -1;
- }
}
hlp->pfds = pfds;
hlp->count = count;
hlp->handle = handle;
+ hlp->mask = mask;
return 0;
}
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- return alsa_poll_helper (alsa->handle, &alsa->pollhlp);
+ return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
}
static int alsa_poll_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
- return alsa_poll_helper (alsa->handle, &alsa->pollhlp);
+ return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
}
static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
+static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
{
switch (fmt) {
case AUD_FMT_S8:
return SND_PCM_FORMAT_U8;
case AUD_FMT_S16:
- return SND_PCM_FORMAT_S16_LE;
+ if (endianness) {
+ return SND_PCM_FORMAT_S16_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_S16_LE;
+ }
case AUD_FMT_U16:
- return SND_PCM_FORMAT_U16_LE;
+ if (endianness) {
+ return SND_PCM_FORMAT_U16_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_U16_LE;
+ }
case AUD_FMT_S32:
- return SND_PCM_FORMAT_S32_LE;
+ if (endianness) {
+ return SND_PCM_FORMAT_S32_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_S32_LE;
+ }
case AUD_FMT_U32:
- return SND_PCM_FORMAT_U32_LE;
+ if (endianness) {
+ return SND_PCM_FORMAT_U32_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_U32_LE;
+ }
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
}
static void alsa_dump_info (struct alsa_params_req *req,
- struct alsa_params_obt *obt)
+ struct alsa_params_obt *obt,
+ snd_pcm_format_t obtfmt)
{
dolog ("parameter | requested value | obtained value\n");
- dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
+ dolog ("format | %10d | %10d\n", req->fmt, obtfmt);
dolog ("channels | %10d | %10d\n",
req->nchannels, obt->nchannels);
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
}
static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep)
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ ALSAConf *conf)
{
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+ const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
- if (err < 0 && conf.verbose) {
+ if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
}
goto err;
}
- if ((req->override_mask & 1) && (obt - req->period_size))
+ if (((req->override_mask & 1) && (obt - req->period_size)))
dolog ("Requested period %s %u was rejected, using %lu\n",
size_in_usec ? "time" : "size", req->period_size, obt);
}
goto err;
}
- if (!in && conf.threshold) {
+ if (!in && conf->threshold) {
snd_pcm_uframes_t threshold;
int bytes_per_sec;
break;
}
- threshold = (conf.threshold * bytes_per_sec) / 1000;
+ threshold = (conf->threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
*handlep = handle;
- if (conf.verbose &&
- (obt->fmt != req->fmt ||
+ if (obtfmt != req->fmt ||
obt->nchannels != req->nchannels ||
- obt->freq != req->freq)) {
- dolog ("Audio paramters for %s\n", typ);
- alsa_dump_info (req, obt);
+ obt->freq != req->freq) {
+ dolog ("Audio parameters for %s\n", typ);
+ alsa_dump_info (req, obt, obtfmt);
}
#ifdef DEBUG
- alsa_dump_info (req, obt);
+ alsa_dump_info (req, obt, obtfmt);
#endif
return 0;
return avail;
}
-static int alsa_run_out (HWVoiceOut *hw)
+static void alsa_write_pending (ALSAVoiceOut *alsa)
{
- ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- int rpos, live, decr;
- int samples;
- uint8_t *dst;
- struct st_sample *src;
- snd_pcm_sframes_t avail;
+ HWVoiceOut *hw = &alsa->hw;
- live = audio_pcm_hw_get_live_out (hw);
- if (!live) {
- return 0;
- }
-
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of available playback frames\n");
- return 0;
- }
-
- decr = audio_MIN (live, avail);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int len = audio_MIN (samples, left_till_end_samples);
- snd_pcm_sframes_t written;
-
- src = hw->mix_buf + rpos;
- dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
-
- hw->clip (dst, src, len);
+ while (alsa->pending) {
+ int left_till_end_samples = hw->samples - alsa->wpos;
+ int len = audio_MIN (alsa->pending, left_till_end_samples);
+ char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
while (len) {
- written = snd_pcm_writei (alsa->handle, dst, len);
+ snd_pcm_sframes_t written;
+
+ written = snd_pcm_writei (alsa->handle, src, len);
if (written <= 0) {
switch (written) {
case 0:
- if (conf.verbose) {
- dolog ("Failed to write %d frames (wrote zero)\n", len);
- }
- goto exit;
+ trace_alsa_wrote_zero(len);
+ return;
case -EPIPE:
if (alsa_recover (alsa->handle)) {
alsa_logerr (written, "Failed to write %d frames\n",
len);
- goto exit;
- }
- if (conf.verbose) {
- dolog ("Recovering from playback xrun\n");
+ return;
}
+ trace_alsa_xrun_out();
continue;
case -ESTRPIPE:
if (alsa_resume (alsa->handle)) {
alsa_logerr (written, "Failed to write %d frames\n",
len);
- goto exit;
- }
- if (conf.verbose) {
- dolog ("Resuming suspended output stream\n");
+ return;
}
+ trace_alsa_resume_out();
continue;
case -EAGAIN:
- goto exit;
+ return;
default:
- alsa_logerr (written, "Failed to write %d frames to %p\n",
- len, dst);
- goto exit;
+ alsa_logerr (written, "Failed to write %d frames from %p\n",
+ len, src);
+ return;
}
}
- rpos = (rpos + written) % hw->samples;
- samples -= written;
+ alsa->wpos = (alsa->wpos + written) % hw->samples;
+ alsa->pending -= written;
len -= written;
- dst = advance (dst, written << hw->info.shift);
- src += written;
}
}
+}
- exit:
- hw->rpos = rpos;
+static int alsa_run_out (HWVoiceOut *hw, int live)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ int decr;
+ snd_pcm_sframes_t avail;
+
+ avail = alsa_get_avail (alsa->handle);
+ if (avail < 0) {
+ dolog ("Could not get number of available playback frames\n");
+ return 0;
+ }
+
+ decr = audio_MIN (live, avail);
+ decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
+ alsa->pending += decr;
+ alsa_write_pending (alsa);
return decr;
}
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
- if (alsa->pcm_buf) {
- qemu_free (alsa->pcm_buf);
- alsa->pcm_buf = NULL;
- }
+ g_free(alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
}
-static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
+ ALSAConf *conf = drv_opaque;
- req.fmt = aud_to_alsafmt (as->fmt);
+ req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf.period_size_out;
- req.buffer_size = conf.buffer_size_out;
- req.size_in_usec = conf.size_in_usec_out;
+ req.period_size = conf->period_size_out;
+ req.buffer_size = conf->buffer_size_out;
+ req.size_in_usec = conf->size_in_usec_out;
req.override_mask =
- (conf.period_size_out_overridden ? 1 : 0) |
- (conf.buffer_size_out_overridden ? 2 : 0);
+ (conf->period_size_out_overridden ? 1 : 0) |
+ (conf->buffer_size_out_overridden ? 2 : 0);
- if (alsa_open (0, &req, &obt, &handle)) {
+ if (alsa_open (0, &req, &obt, &handle, conf)) {
return -1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
- alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
+ alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
-static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
+#define VOICE_CTL_PAUSE 0
+#define VOICE_CTL_PREPARE 1
+#define VOICE_CTL_START 2
+
+static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
{
int err;
- if (pause) {
+ if (ctl == VOICE_CTL_PAUSE) {
err = snd_pcm_drop (handle);
if (err < 0) {
alsa_logerr (err, "Could not stop %s\n", typ);
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
return -1;
}
+ if (ctl == VOICE_CTL_START) {
+ err = snd_pcm_start(handle);
+ if (err < 0) {
+ alsa_logerr (err, "Could not start handle for %s\n", typ);
+ return -1;
+ }
+ }
}
return 0;
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
- va_list ap;
- int poll_mode;
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
-
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_out (hw)) {
- poll_mode = 0;
+ {
+ va_list ap;
+ int poll_mode;
+
+ va_start (ap, cmd);
+ poll_mode = va_arg (ap, int);
+ va_end (ap);
+
+ ldebug ("enabling voice\n");
+ if (poll_mode && alsa_poll_out (hw)) {
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
+ return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
}
- hw->poll_mode = poll_mode;
- return alsa_voice_ctl (alsa->handle, "playback", 0);
case VOICE_DISABLE:
ldebug ("disabling voice\n");
- return alsa_voice_ctl (alsa->handle, "playback", 1);
+ if (hw->poll_mode) {
+ hw->poll_mode = 0;
+ alsa_fini_poll (&alsa->pollhlp);
+ }
+ return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
}
return -1;
}
-static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
+ ALSAConf *conf = drv_opaque;
- req.fmt = aud_to_alsafmt (as->fmt);
+ req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf.period_size_in;
- req.buffer_size = conf.buffer_size_in;
- req.size_in_usec = conf.size_in_usec_in;
+ req.period_size = conf->period_size_in;
+ req.buffer_size = conf->buffer_size_in;
+ req.size_in_usec = conf->size_in_usec_in;
req.override_mask =
- (conf.period_size_in_overridden ? 1 : 0) |
- (conf.buffer_size_in_overridden ? 2 : 0);
+ (conf->period_size_in_overridden ? 1 : 0) |
+ (conf->buffer_size_in_overridden ? 2 : 0);
- if (alsa_open (1, &req, &obt, &handle)) {
+ if (alsa_open (1, &req, &obt, &handle, conf)) {
return -1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
- alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+ alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
- if (alsa->pcm_buf) {
- qemu_free (alsa->pcm_buf);
- alsa->pcm_buf = NULL;
- }
+ g_free(alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
}
static int alsa_run_in (HWVoiceIn *hw)
dolog ("Failed to resume suspended input stream\n");
return 0;
}
- if (conf.verbose) {
- dolog ("Resuming suspended input stream\n");
- }
+ trace_alsa_resume_in();
break;
default:
- if (conf.verbose) {
- dolog ("No frames available and ALSA state is %d\n", state);
- }
+ trace_alsa_no_frames(state);
return 0;
}
}
if (nread <= 0) {
switch (nread) {
case 0:
- if (conf.verbose) {
- dolog ("Failed to read %ld frames (read zero)\n", len);
- }
+ trace_alsa_read_zero(len);
goto exit;
case -EPIPE:
alsa_logerr (nread, "Failed to read %ld frames\n", len);
goto exit;
}
- if (conf.verbose) {
- dolog ("Recovering from capture xrun\n");
- }
+ trace_alsa_xrun_in();
continue;
case -EAGAIN:
}
}
- hw->conv (dst, src, nread, &nominal_volume);
+ hw->conv (dst, src, nread);
src = advance (src, nread << hwshift);
dst += nread;
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
- va_list ap;
- int poll_mode;
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
-
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_in (hw)) {
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
+ {
+ va_list ap;
+ int poll_mode;
+
+ va_start (ap, cmd);
+ poll_mode = va_arg (ap, int);
+ va_end (ap);
+
+ ldebug ("enabling voice\n");
+ if (poll_mode && alsa_poll_in (hw)) {
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
- return alsa_voice_ctl (alsa->handle, "capture", 0);
+ return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
+ }
case VOICE_DISABLE:
ldebug ("disabling voice\n");
hw->poll_mode = 0;
alsa_fini_poll (&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "capture", 1);
+ return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
}
return -1;
}
+static ALSAConf glob_conf = {
+ .buffer_size_out = 4096,
+ .period_size_out = 1024,
+ .pcm_name_out = "default",
+ .pcm_name_in = "default",
+};
+
static void *alsa_audio_init (void)
{
- return &conf;
+ ALSAConf *conf = g_malloc(sizeof(ALSAConf));
+ *conf = glob_conf;
+ return conf;
}
static void alsa_audio_fini (void *opaque)
{
- (void) opaque;
+ g_free(opaque);
}
static struct audio_option alsa_options[] = {
{
.name = "DAC_SIZE_IN_USEC",
.tag = AUD_OPT_BOOL,
- .valp = &conf.size_in_usec_out,
+ .valp = &glob_conf.size_in_usec_out,
.descr = "DAC period/buffer size in microseconds (otherwise in frames)"
},
{
.name = "DAC_PERIOD_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.period_size_out,
+ .valp = &glob_conf.period_size_out,
.descr = "DAC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_out_overridden
+ .overriddenp = &glob_conf.period_size_out_overridden
},
{
.name = "DAC_BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_out,
+ .valp = &glob_conf.buffer_size_out,
.descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_out_overridden
+ .overriddenp = &glob_conf.buffer_size_out_overridden
},
{
.name = "ADC_SIZE_IN_USEC",
.tag = AUD_OPT_BOOL,
- .valp = &conf.size_in_usec_in,
+ .valp = &glob_conf.size_in_usec_in,
.descr =
"ADC period/buffer size in microseconds (otherwise in frames)"
},
{
.name = "ADC_PERIOD_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.period_size_in,
+ .valp = &glob_conf.period_size_in,
.descr = "ADC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_in_overridden
+ .overriddenp = &glob_conf.period_size_in_overridden
},
{
.name = "ADC_BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_in,
+ .valp = &glob_conf.buffer_size_in,
.descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_in_overridden
+ .overriddenp = &glob_conf.buffer_size_in_overridden
},
{
.name = "THRESHOLD",
.tag = AUD_OPT_INT,
- .valp = &conf.threshold,
+ .valp = &glob_conf.threshold,
.descr = "(undocumented)"
},
{
.name = "DAC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_out,
+ .valp = &glob_conf.pcm_name_out,
.descr = "DAC device name (for instance dmix)"
},
{
.name = "ADC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_in,
+ .valp = &glob_conf.pcm_name_in,
.descr = "ADC device name"
},
- {
- .name = "VERBOSE",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.verbose,
- .descr = "Behave in a more verbose way"
- },
{ /* End of list */ }
};
.ctl_in = alsa_ctl_in,
};
-struct audio_driver alsa_audio_driver = {
+static struct audio_driver alsa_audio_driver = {
.name = "alsa",
.descr = "ALSA http://www.alsa-project.org",
.options = alsa_options,
.voice_size_out = sizeof (ALSAVoiceOut),
.voice_size_in = sizeof (ALSAVoiceIn)
};
+
+static void register_audio_alsa(void)
+{
+ audio_driver_register(&alsa_audio_driver);
+}
+type_init(register_audio_alsa);
projects / qemu.git / blobdiff