* 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/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 ALSAVoiceOut {
- HWVoiceOut hw;
- void *pcm_buf;
- snd_pcm_t *handle;
-} ALSAVoiceOut;
-
-typedef struct ALSAVoiceIn {
- HWVoiceIn hw;
- snd_pcm_t *handle;
- void *pcm_buf;
-} ALSAVoiceIn;
-
-static struct {
+typedef struct ALSAConf {
int size_in_usec_in;
int size_in_usec_out;
const char *pcm_name_in;
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",
+} 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;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+ HWVoiceIn hw;
+ snd_pcm_t *handle;
+ void *pcm_buf;
+ struct pollhlp pollhlp;
+} ALSAVoiceIn;
+
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}
-static void alsa_anal_close (snd_pcm_t **handlep)
+static void alsa_fini_poll (struct pollhlp *hlp)
+{
+ int i;
+ struct pollfd *pfds = hlp->pfds;
+
+ if (pfds) {
+ for (i = 0; i < hlp->count; ++i) {
+ qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
+ }
+ g_free (pfds);
+ }
+ hlp->pfds = NULL;
+ hlp->count = 0;
+ hlp->handle = NULL;
+}
+
+static void alsa_anal_close1 (snd_pcm_t **handlep)
{
int err = snd_pcm_close (*handlep);
if (err) {
*handlep = NULL;
}
+static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
+{
+ alsa_fini_poll (hlp);
+ alsa_anal_close1 (handlep);
+}
+
+static int alsa_recover (snd_pcm_t *handle)
+{
+ int err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to prepare handle %p\n", handle);
+ return -1;
+ }
+ return 0;
+}
+
+static int alsa_resume (snd_pcm_t *handle)
+{
+ int err = snd_pcm_resume (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to resume handle %p\n", handle);
+ return -1;
+ }
+ return 0;
+}
+
+static void alsa_poll_handler (void *opaque)
+{
+ int err, count;
+ snd_pcm_state_t state;
+ struct pollhlp *hlp = opaque;
+ unsigned short revents;
+
+ count = poll (hlp->pfds, hlp->count, 0);
+ if (count < 0) {
+ dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
+ return;
+ }
+
+ if (!count) {
+ return;
+ }
+
+ /* XXX: ALSA example uses initial count, not the one returned by
+ poll, correct? */
+ err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
+ hlp->count, &revents);
+ if (err < 0) {
+ alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
+ return;
+ }
+
+ 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;
+
+ case SND_PCM_STATE_SUSPENDED:
+ alsa_resume (hlp->handle);
+ break;
+
+ case SND_PCM_STATE_PREPARED:
+ audio_run ("alsa run (prepared)");
+ break;
+
+ case SND_PCM_STATE_RUNNING:
+ audio_run ("alsa run (running)");
+ break;
+
+ default:
+ dolog ("Unexpected state %d\n", state);
+ }
+}
+
+static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
+{
+ int i, count, err;
+ struct pollfd *pfds;
+
+ count = snd_pcm_poll_descriptors_count (handle);
+ if (count <= 0) {
+ dolog ("Could not initialize poll mode\n"
+ "Invalid number of poll descriptors %d\n", count);
+ return -1;
+ }
+
+ pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
+ if (!pfds) {
+ dolog ("Could not initialize poll mode\n");
+ return -1;
+ }
+
+ err = snd_pcm_poll_descriptors (handle, pfds, count);
+ if (err < 0) {
+ alsa_logerr (err, "Could not initialize poll mode\n"
+ "Could not obtain poll descriptors\n");
+ g_free (pfds);
+ return -1;
+ }
+
+ for (i = 0; i < count; ++i) {
+ if (pfds[i].events & POLLIN) {
+ qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
+ }
+ if (pfds[i].events & POLLOUT) {
+ trace_alsa_pollout(i, pfds[i].fd);
+ qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
+ }
+ trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
+
+ }
+ hlp->pfds = pfds;
+ hlp->count = count;
+ hlp->handle = handle;
+ hlp->mask = mask;
+ return 0;
+}
+
+static int alsa_poll_out (HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ 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, 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;
err:
- alsa_anal_close (&handle);
+ alsa_anal_close1 (&handle);
return -1;
}
-static int alsa_recover (snd_pcm_t *handle)
-{
- int err = snd_pcm_prepare (handle);
- if (err < 0) {
- alsa_logerr (err, "Failed to prepare handle %p\n", handle);
- return -1;
- }
- return 0;
-}
-
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
{
snd_pcm_sframes_t avail;
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;
- st_sample_t *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;
+ return;
}
- if (conf.verbose) {
- dolog ("Recovering from playback xrun\n");
+ trace_alsa_xrun_out();
+ continue;
+
+ case -ESTRPIPE:
+ /* stream is suspended and waiting for an
+ application recovery */
+ if (alsa_resume (alsa->handle)) {
+ alsa_logerr (written, "Failed to write %d frames\n",
+ len);
+ 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;
}
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
ldebug ("alsa_fini\n");
- alsa_anal_close (&alsa->handle);
+ 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, audsettings_t *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;
- audsettings_t obt_as;
+ 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.override_mask = !!conf.period_size_out_overridden
- | (!!conf.buffer_size_out_overridden << 1);
-
- if (alsa_open (0, &req, &obt, &handle)) {
+ 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);
+
+ if (alsa_open (0, &req, &obt, &handle, conf)) {
return -1;
}
if (!alsa->pcm_buf) {
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
- alsa_anal_close (&handle);
+ alsa_anal_close1 (&handle);
return -1;
}
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;
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- return alsa_voice_ctl (alsa->handle, "playback", 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);
+ }
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, audsettings_t *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;
- audsettings_t obt_as;
+ 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.override_mask = !!conf.period_size_in_overridden
- | (!!conf.buffer_size_in_overridden << 1);
-
- if (alsa_open (1, &req, &obt, &handle)) {
+ 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);
+
+ if (alsa_open (1, &req, &obt, &handle, conf)) {
return -1;
}
if (!alsa->pcm_buf) {
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
- alsa_anal_close (&handle);
+ alsa_anal_close1 (&handle);
return -1;
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
- alsa_anal_close (&alsa->handle);
+ 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)
int add;
int len;
} bufs[2] = {
- { hw->wpos, 0 },
- { 0, 0 }
+ { .add = hw->wpos, .len = 0 },
+ { .add = 0, .len = 0 }
};
snd_pcm_sframes_t avail;
snd_pcm_uframes_t read_samples = 0;
return 0;
}
- if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
- avail = hw->samples;
+ if (!avail) {
+ snd_pcm_state_t state;
+
+ state = snd_pcm_state (alsa->handle);
+ switch (state) {
+ case SND_PCM_STATE_PREPARED:
+ avail = hw->samples;
+ break;
+ case SND_PCM_STATE_SUSPENDED:
+ /* stream is suspended and waiting for an application recovery */
+ if (alsa_resume (alsa->handle)) {
+ dolog ("Failed to resume suspended input stream\n");
+ return 0;
+ }
+ trace_alsa_resume_in();
+ break;
+ default:
+ trace_alsa_no_frames(state);
+ return 0;
+ }
}
decr = audio_MIN (dead, avail);
for (i = 0; i < 2; ++i) {
void *src;
- st_sample_t *dst;
+ struct st_sample *dst;
snd_pcm_sframes_t nread;
snd_pcm_uframes_t len;
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;
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- return alsa_voice_ctl (alsa->handle, "capture", 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_in (hw)) {
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
+
+ return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
+ }
case VOICE_DISABLE:
ldebug ("disabling voice\n");
- return alsa_voice_ctl (alsa->handle, "capture", 1);
+ if (hw->poll_mode) {
+ hw->poll_mode = 0;
+ alsa_fini_poll (&alsa->pollhlp);
+ }
+ 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[] = {
- {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
- "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
- {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
- "DAC period size (0 to go with system default)",
- &conf.period_size_out_overridden, 0},
- {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
- "DAC buffer size (0 to go with system default)",
- &conf.buffer_size_out_overridden, 0},
-
- {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
- "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
- {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
- "ADC period size (0 to go with system default)",
- &conf.period_size_in_overridden, 0},
- {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
- "ADC buffer size (0 to go with system default)",
- &conf.buffer_size_in_overridden, 0},
-
- {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
- "(undocumented)", NULL, 0},
-
- {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
- "DAC device name (for instance dmix)", NULL, 0},
-
- {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
- "ADC device name", NULL, 0},
-
- {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
- "Behave in a more verbose way", NULL, 0},
-
- {NULL, 0, NULL, NULL, NULL, 0}
+ {
+ .name = "DAC_SIZE_IN_USEC",
+ .tag = AUD_OPT_BOOL,
+ .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 = &glob_conf.period_size_out,
+ .descr = "DAC period size (0 to go with system default)",
+ .overriddenp = &glob_conf.period_size_out_overridden
+ },
+ {
+ .name = "DAC_BUFFER_SIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.buffer_size_out,
+ .descr = "DAC buffer size (0 to go with system default)",
+ .overriddenp = &glob_conf.buffer_size_out_overridden
+ },
+ {
+ .name = "ADC_SIZE_IN_USEC",
+ .tag = AUD_OPT_BOOL,
+ .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 = &glob_conf.period_size_in,
+ .descr = "ADC period size (0 to go with system default)",
+ .overriddenp = &glob_conf.period_size_in_overridden
+ },
+ {
+ .name = "ADC_BUFFER_SIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.buffer_size_in,
+ .descr = "ADC buffer size (0 to go with system default)",
+ .overriddenp = &glob_conf.buffer_size_in_overridden
+ },
+ {
+ .name = "THRESHOLD",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.threshold,
+ .descr = "(undocumented)"
+ },
+ {
+ .name = "DAC_DEV",
+ .tag = AUD_OPT_STR,
+ .valp = &glob_conf.pcm_name_out,
+ .descr = "DAC device name (for instance dmix)"
+ },
+ {
+ .name = "ADC_DEV",
+ .tag = AUD_OPT_STR,
+ .valp = &glob_conf.pcm_name_in,
+ .descr = "ADC device name"
+ },
+ { /* End of list */ }
};
-static const struct audio_pcm_ops alsa_pcm_ops = {
- alsa_init_out,
- alsa_fini_out,
- alsa_run_out,
- alsa_write,
- alsa_ctl_out,
-
- alsa_init_in,
- alsa_fini_in,
- alsa_run_in,
- alsa_read,
- alsa_ctl_in
+static struct audio_pcm_ops alsa_pcm_ops = {
+ .init_out = alsa_init_out,
+ .fini_out = alsa_fini_out,
+ .run_out = alsa_run_out,
+ .write = alsa_write,
+ .ctl_out = alsa_ctl_out,
+
+ .init_in = alsa_init_in,
+ .fini_in = alsa_fini_in,
+ .run_in = alsa_run_in,
+ .read = alsa_read,
+ .ctl_in = alsa_ctl_in,
};
struct audio_driver alsa_audio_driver = {
- INIT_FIELD (name = ) "alsa",
- INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
- INIT_FIELD (options = ) alsa_options,
- INIT_FIELD (init = ) alsa_audio_init,
- INIT_FIELD (fini = ) alsa_audio_fini,
- INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
- INIT_FIELD (can_be_default = ) 1,
- INIT_FIELD (max_voices_out = ) INT_MAX,
- INIT_FIELD (max_voices_in = ) INT_MAX,
- INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
- INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
+ .name = "alsa",
+ .descr = "ALSA http://www.alsa-project.org",
+ .options = alsa_options,
+ .init = alsa_audio_init,
+ .fini = alsa_audio_fini,
+ .pcm_ops = &alsa_pcm_ops,
+ .can_be_default = 1,
+ .max_voices_out = INT_MAX,
+ .max_voices_in = INT_MAX,
+ .voice_size_out = sizeof (ALSAVoiceOut),
+ .voice_size_in = sizeof (ALSAVoiceIn)
};
projects / mirror_qemu.git / blobdiff