* 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 ALSAVoiceOut {
HWVoiceOut hw;
+ int wpos;
+ int pending;
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
snd_pcm_t *handle;
void *pcm_buf;
struct pollhlp pollhlp;
+ Audiodev *dev;
} 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;
int nchannels;
- int size_in_usec;
- int override_mask;
- unsigned int buffer_size;
- unsigned int period_size;
};
struct alsa_params_obt {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
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;
}
if (!(revents & hlp->mask)) {
- if (conf.verbose) {
- dolog ("revents = %d\n", revents);
- }
+ 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;
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;
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 (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
- case AUD_FMT_S16:
- return SND_PCM_FORMAT_S16_LE;
+ case AUDIO_FORMAT_S16:
+ 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;
+ case AUDIO_FORMAT_U16:
+ 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;
+ case AUDIO_FORMAT_S32:
+ 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;
+ case AUDIO_FORMAT_U32:
+ 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 int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
default:
}
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,
+ AudiodevAlsaPerDirectionOptions *apdo)
{
- dolog ("parameter | requested value | obtained value\n");
- dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
- dolog ("channels | %10d | %10d\n",
- req->nchannels, obt->nchannels);
- dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
- dolog ("============================================\n");
- dolog ("requested: buffer size %d period size %d\n",
- req->buffer_size, req->period_size);
- dolog ("obtained: samples %ld\n", obt->samples);
+ dolog("parameter | requested value | obtained value\n");
+ 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);
+ dolog("============================================\n");
+ dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
+ apdo->buffer_length, apdo->period_length);
+ dolog("obtained: samples %ld\n", obt->samples);
}
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
}
}
-static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep)
+static int alsa_open(bool in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ Audiodev *dev)
{
+ AudiodevAlsaOptions *aopts = &dev->u.alsa;
+ AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
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 = apdo->has_dev ? apdo->dev : "default";
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
nchannels = req->nchannels;
- size_in_usec = req->size_in_usec;
snd_pcm_hw_params_alloca (&hw_params);
}
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->buffer_size) {
- unsigned long obt;
+ if (apdo->buffer_length) {
+ int dir = 0;
+ unsigned int btime = apdo->buffer_length;
- if (size_in_usec) {
- int dir = 0;
- unsigned int btime = req->buffer_size;
+ err = snd_pcm_hw_params_set_buffer_time_near(
+ handle, hw_params, &btime, &dir);
- err = snd_pcm_hw_params_set_buffer_time_near (
- handle,
- hw_params,
- &btime,
- &dir
- );
- obt = btime;
- }
- else {
- snd_pcm_uframes_t bsize = req->buffer_size;
-
- err = snd_pcm_hw_params_set_buffer_size_near (
- handle,
- hw_params,
- &bsize
- );
- obt = bsize;
- }
if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
- size_in_usec ? "time" : "size", req->buffer_size);
+ alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
+ apdo->buffer_length);
goto err;
}
- if ((req->override_mask & 2) && (obt - req->buffer_size))
- dolog ("Requested buffer %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->buffer_size, obt);
+ if (apdo->has_buffer_length && btime != apdo->buffer_length) {
+ dolog("Requested buffer time %" PRId32
+ " was rejected, using %u\n", apdo->buffer_length, btime);
+ }
}
- if (req->period_size) {
- unsigned long obt;
-
- if (size_in_usec) {
- int dir = 0;
- unsigned int ptime = req->period_size;
+ if (apdo->period_length) {
+ int dir = 0;
+ unsigned int ptime = apdo->period_length;
- err = snd_pcm_hw_params_set_period_time_near (
- handle,
- hw_params,
- &ptime,
- &dir
- );
- obt = ptime;
- }
- else {
- int dir = 0;
- snd_pcm_uframes_t psize = req->period_size;
-
- err = snd_pcm_hw_params_set_period_size_near (
- handle,
- hw_params,
- &psize,
- &dir
- );
- obt = psize;
- }
+ err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
+ &dir);
if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
- size_in_usec ? "time" : "size", req->period_size);
+ alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
+ apdo->period_length);
goto err;
}
- 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);
+ if (apdo->has_period_length && ptime != apdo->period_length) {
+ dolog("Requested period time %" PRId32 " was rejected, using %d\n",
+ apdo->period_length, ptime);
+ }
}
err = snd_pcm_hw_params (handle, hw_params);
goto err;
}
- if (!in && conf.threshold) {
- snd_pcm_uframes_t threshold;
- int bytes_per_sec;
-
- bytes_per_sec = freq << (nchannels == 2);
-
- switch (obt->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- break;
-
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- bytes_per_sec <<= 1;
- break;
-
- case AUD_FMT_S32:
- case AUD_FMT_U32:
- bytes_per_sec <<= 2;
- break;
- }
-
- threshold = (conf.threshold * bytes_per_sec) / 1000;
- alsa_set_threshold (handle, threshold);
+ if (!in && aopts->has_threshold && aopts->threshold) {
+ struct audsettings as = { .freq = freq };
+ alsa_set_threshold(
+ handle,
+ audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
+ &as, aopts->threshold));
}
obt->nchannels = nchannels;
*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, apdo);
}
#ifdef DEBUG
- alsa_dump_info (req, obt);
+ alsa_dump_info(req, obt, obtfmt, pdo);
#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;
+ Audiodev *dev = 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 ? 1 : 0) |
- (conf.buffer_size_out_overridden ? 2 : 0);
-
- if (alsa_open (0, &req, &obt, &handle)) {
+
+ if (alsa_open(0, &req, &obt, &handle, dev)) {
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->dev = dev;
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);
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_out (hw)) {
- poll_mode = 0;
+ {
+ bool poll_mode = apdo->try_poll;
+
+ 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;
+ Audiodev *dev = 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 ? 1 : 0) |
- (conf.buffer_size_in_overridden ? 2 : 0);
-
- if (alsa_open (1, &req, &obt, &handle)) {
+
+ if (alsa_open(1, &req, &obt, &handle, dev)) {
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->dev = dev;
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);
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_in (hw)) {
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
+ {
+ bool poll_mode = apdo->try_poll;
- return alsa_voice_ctl (alsa->handle, "capture", 0);
+ 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");
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 void *alsa_audio_init (void)
+static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
{
- return &conf;
+ if (!apdo->has_try_poll) {
+ apdo->try_poll = true;
+ apdo->has_try_poll = true;
+ }
}
-static void alsa_audio_fini (void *opaque)
+static void *alsa_audio_init(Audiodev *dev)
{
- (void) opaque;
+ AudiodevAlsaOptions *aopts;
+ assert(dev->driver == AUDIODEV_DRIVER_ALSA);
+
+ aopts = &dev->u.alsa;
+ alsa_init_per_direction(aopts->in);
+ alsa_init_per_direction(aopts->out);
+
+ /*
+ * need to define them, as otherwise alsa produces no sound
+ * doesn't set has_* so alsa_open can identify it wasn't set by the user
+ */
+ if (!dev->u.alsa.out->has_period_length) {
+ /* 1024 frames assuming 44100Hz */
+ dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
+ }
+ if (!dev->u.alsa.out->has_buffer_length) {
+ /* 4096 frames assuming 44100Hz */
+ dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
+ }
+
+ /*
+ * OptsVisitor sets unspecified optional fields to zero, but do not depend
+ * on it...
+ */
+ if (!dev->u.alsa.in->has_period_length) {
+ dev->u.alsa.in->period_length = 0;
+ }
+ if (!dev->u.alsa.in->has_buffer_length) {
+ dev->u.alsa.in->buffer_length = 0;
+ }
+
+ return dev;
}
-static struct audio_option alsa_options[] = {
- {
- .name = "DAC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &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,
- .descr = "DAC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_out_overridden
- },
- {
- .name = "DAC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_out,
- .descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_out_overridden
- },
- {
- .name = "ADC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &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,
- .descr = "ADC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_in_overridden
- },
- {
- .name = "ADC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_in,
- .descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_in_overridden
- },
- {
- .name = "THRESHOLD",
- .tag = AUD_OPT_INT,
- .valp = &conf.threshold,
- .descr = "(undocumented)"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_out,
- .descr = "DAC device name (for instance dmix)"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &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 */ }
-};
+static void alsa_audio_fini (void *opaque)
+{
+}
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.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,
.init = alsa_audio_init,
.fini = alsa_audio_fini,
.pcm_ops = &alsa_pcm_ops,
.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 / mirror_qemu.git / blobdiff