* THE SOFTWARE.
*/
#include <alsa/asoundlib.h>
-#include "vl.h"
+#include "qemu-common.h"
+#include "qemu/main-loop.h"
+#include "audio.h"
+
+#if QEMU_GNUC_PREREQ(4, 3)
+#pragma GCC diagnostic ignored "-Waddress"
+#endif
#define AUDIO_CAP "alsa"
#include "audio_int.h"
+struct pollhlp {
+ snd_pcm_t *handle;
+ struct pollfd *pfds;
+ int count;
+ int mask;
+};
+
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
+ int wpos;
+ int pending;
void *pcm_buf;
snd_pcm_t *handle;
- int can_pause;
- int was_enabled;
+ struct pollhlp pollhlp;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
void *pcm_buf;
- int can_pause;
+ struct pollhlp pollhlp;
} ALSAVoiceIn;
static struct {
unsigned int period_size_out;
unsigned int threshold;
- int buffer_size_in_overriden;
- int period_size_in_overriden;
+ int buffer_size_in_overridden;
+ int period_size_in_overridden;
- int buffer_size_out_overriden;
- int period_size_out_overriden;
+ int buffer_size_out_overridden;
+ int period_size_out_overridden;
+ int verbose;
} conf = {
-#ifdef HIGH_LATENCY
- .size_in_usec_in = 1,
- .size_in_usec_out = 1,
-#endif
- .pcm_name_out = "hw:0,0",
- .pcm_name_in = "hw:0,0",
-#ifdef HIGH_LATENCY
- .buffer_size_in = 400000,
- .period_size_in = 400000 / 4,
- .buffer_size_out = 400000,
- .period_size_out = 400000 / 4,
-#else
-#define DEFAULT_BUFFER_SIZE 1024
-#define DEFAULT_PERIOD_SIZE 256
- .buffer_size_in = DEFAULT_BUFFER_SIZE,
- .period_size_in = DEFAULT_PERIOD_SIZE,
- .buffer_size_out = DEFAULT_BUFFER_SIZE,
- .period_size_out = DEFAULT_PERIOD_SIZE,
- .buffer_size_in_overriden = 0,
- .buffer_size_out_overriden = 0,
- .period_size_in_overriden = 0,
- .period_size_out_overriden = 0,
-#endif
- .threshold = 0
+ .buffer_size_out = 4096,
+ .period_size_out = 1024,
+ .pcm_name_out = "default",
+ .pcm_name_in = "default",
};
struct alsa_params_req {
int freq;
- audfmt_e fmt;
+ 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;
+ int endianness;
int nchannels;
- int can_pause;
snd_pcm_uframes_t samples;
};
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)) {
+ if (conf.verbose) {
+ dolog ("revents = %d\n", 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) {
+ err = 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);
+ }
+ 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);
+
+ while (i--) {
+ qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
+ }
+ g_free (pfds);
+ return -1;
+ }
+ }
+ 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 int 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:
+ if (endianness) {
+ return SND_PCM_FORMAT_S32_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_S32_LE;
+ }
+
+ case AUD_FMT_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 (int alsafmt, audfmt_e *fmt, int *endianness)
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+ int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*fmt = AUD_FMT_U16;
break;
+ case SND_PCM_FORMAT_S32_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_S32;
+ break;
+
+ case SND_PCM_FORMAT_U32_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_U32;
+ break;
+
+ case SND_PCM_FORMAT_S32_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_S32;
+ break;
+
+ case SND_PCM_FORMAT_U32_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_U32;
+ break;
+
default:
dolog ("Unrecognized audio format %d\n", alsafmt);
return -1;
return 0;
}
-#if defined DEBUG_MISMATCHES || defined DEBUG
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);
req->buffer_size, req->period_size);
dolog ("obtained: samples %ld\n", obt->samples);
}
-#endif
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
{
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
- int err, freq, nchannels;
+ int err;
+ int size_in_usec;
+ unsigned int freq, nchannels;
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
- unsigned int period_size, buffer_size;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
+ snd_pcm_format_t obtfmt;
freq = req->freq;
- period_size = req->period_size;
- buffer_size = req->buffer_size;
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) {
+ if (err < 0 && conf.verbose) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
- goto err;
}
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
goto err;
}
- if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
- if (!buffer_size) {
- buffer_size = DEFAULT_BUFFER_SIZE;
- period_size= DEFAULT_PERIOD_SIZE;
- }
- }
+ if (req->buffer_size) {
+ unsigned long obt;
- if (buffer_size) {
- if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
- if (period_size) {
- err = snd_pcm_hw_params_set_period_time_near (
- handle,
- hw_params,
- &period_size,
- 0
- );
- if (err < 0) {
- alsa_logerr2 (err, typ,
- "Failed to set period time %d\n",
- req->period_size);
- goto err;
- }
- }
+ 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,
- &buffer_size,
- 0
+ &btime,
+ &dir
);
-
- if (err < 0) {
- alsa_logerr2 (err, typ,
- "Failed to set buffer time %d\n",
- req->buffer_size);
- goto err;
- }
+ obt = btime;
}
else {
- int dir;
- snd_pcm_uframes_t minval;
-
- if (period_size) {
- minval = period_size;
- dir = 0;
-
- err = snd_pcm_hw_params_get_period_size_min (
- hw_params,
- &minval,
- &dir
- );
- if (err < 0) {
- alsa_logerr (
- err,
- "Could not get minmal period size for %s\n",
- typ
- );
- }
- else {
- if (period_size < minval) {
- if ((in && conf.period_size_in_overriden)
- || (!in && conf.period_size_out_overriden)) {
- dolog ("%s period size(%d) is less "
- "than minmal period size(%ld)\n",
- typ,
- period_size,
- minval);
- }
- period_size = minval;
- }
- }
+ snd_pcm_uframes_t bsize = req->buffer_size;
- err = snd_pcm_hw_params_set_period_size (
- handle,
- hw_params,
- period_size,
- 0
- );
- if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set period size %d\n",
- req->period_size);
- goto err;
- }
- }
+ 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);
+ 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);
+ }
- minval = buffer_size;
- err = snd_pcm_hw_params_get_buffer_size_min (
+ if (req->period_size) {
+ unsigned long obt;
+
+ if (size_in_usec) {
+ int dir = 0;
+ unsigned int ptime = req->period_size;
+
+ err = snd_pcm_hw_params_set_period_time_near (
+ handle,
hw_params,
- &minval
+ &ptime,
+ &dir
);
- if (err < 0) {
- alsa_logerr (err, "Could not get minmal buffer size for %s\n",
- typ);
- }
- else {
- if (buffer_size < minval) {
- if ((in && conf.buffer_size_in_overriden)
- || (!in && conf.buffer_size_out_overriden)) {
- dolog (
- "%s buffer size(%d) is less "
- "than minimal buffer size(%ld)\n",
- typ,
- buffer_size,
- minval
- );
- }
- buffer_size = minval;
- }
- }
+ obt = ptime;
+ }
+ else {
+ int dir = 0;
+ snd_pcm_uframes_t psize = req->period_size;
- err = snd_pcm_hw_params_set_buffer_size (
+ err = snd_pcm_hw_params_set_period_size_near (
handle,
hw_params,
- buffer_size
+ &psize,
+ &dir
);
- if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
- req->buffer_size);
- goto err;
- }
+ obt = psize;
}
- }
- else {
- dolog ("warning: Buffer size is not set\n");
+
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
+ size_in_usec ? "time" : "size", req->period_size);
+ 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);
}
err = snd_pcm_hw_params (handle, hw_params);
goto err;
}
- err = snd_pcm_prepare (handle);
+ err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
if (err < 0) {
- alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
+ alsa_logerr2 (err, typ, "Failed to get format\n");
goto err;
}
- obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
- if (obt->can_pause < 0) {
- alsa_logerr (err, "Could not get pause capability for %s\n", typ);
- obt->can_pause = 0;
+ if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
+ dolog ("Invalid format was returned %d\n", obtfmt);
+ goto err;
+ }
+
+ err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
+ goto err;
}
if (!in && conf.threshold) {
snd_pcm_uframes_t threshold;
int bytes_per_sec;
- bytes_per_sec = freq
- << (nchannels == 2)
- << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
+ 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);
}
- obt->fmt = req->fmt;
obt->nchannels = nchannels;
obt->freq = freq;
obt->samples = obt_buffer_size;
+
*handlep = handle;
-#if defined DEBUG_MISMATCHES || defined DEBUG
- if (obt->fmt != req->fmt ||
- obt->nchannels != req->nchannels ||
- obt->freq != req->freq) {
- dolog ("Audio paramters mismatch for %s\n", typ);
- alsa_dump_info (req, obt);
+ if (conf.verbose &&
+ (obtfmt != req->fmt ||
+ obt->nchannels != req->nchannels ||
+ obt->freq != req->freq)) {
+ dolog ("Audio parameters for %s\n", typ);
+ alsa_dump_info (req, obt, obtfmt);
}
-#endif
#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 int alsa_run_out (HWVoiceOut *hw)
+static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
{
- ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- int rpos, live, decr;
- int samples;
- uint8_t *dst;
- st_sample_t *src;
snd_pcm_sframes_t avail;
- live = audio_pcm_hw_get_live_out (hw);
- if (!live) {
- return 0;
- }
-
- avail = snd_pcm_avail_update (alsa->handle);
+ avail = snd_pcm_avail_update (handle);
if (avail < 0) {
if (avail == -EPIPE) {
- if (!alsa_recover (alsa->handle)) {
- avail = snd_pcm_avail_update (alsa->handle);
- if (avail >= 0) {
- goto ok;
- }
+ if (!alsa_recover (handle)) {
+ avail = snd_pcm_avail_update (handle);
}
}
- alsa_logerr (avail, "Could not get amount free space\n");
- return 0;
+ if (avail < 0) {
+ alsa_logerr (avail,
+ "Could not obtain number of available frames\n");
+ return -1;
+ }
}
- ok:
- decr = audio_MIN (live, avail);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int convert_samples = 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, convert_samples);
-
- again:
- written = snd_pcm_writei (alsa->handle, dst, convert_samples);
-
- if (written < 0) {
- switch (written) {
- case -EPIPE:
- if (!alsa_recover (alsa->handle)) {
- goto again;
+ return avail;
+}
+
+static void alsa_write_pending (ALSAVoiceOut *alsa)
+{
+ HWVoiceOut *hw = &alsa->hw;
+
+ 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) {
+ 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);
+ }
+ return;
+
+ case -EPIPE:
+ if (alsa_recover (alsa->handle)) {
+ alsa_logerr (written, "Failed to write %d frames\n",
+ len);
+ return;
+ }
+ if (conf.verbose) {
+ dolog ("Recovering from playback xrun\n");
+ }
+ 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;
+ }
+ if (conf.verbose) {
+ dolog ("Resuming suspended output stream\n");
+ }
+ continue;
+
+ case -EAGAIN:
+ return;
+
+ default:
+ alsa_logerr (written, "Failed to write %d frames from %p\n",
+ len, src);
+ return;
}
- dolog (
- "Failed to write %d frames to %p, handle %p not prepared\n",
- convert_samples,
- dst,
- alsa->handle
- );
- goto exit;
-
- case -EAGAIN:
- goto again;
-
- default:
- alsa_logerr (written, "Failed to write %d frames to %p\n",
- convert_samples, dst);
- goto exit;
}
+
+ alsa->wpos = (alsa->wpos + written) % hw->samples;
+ alsa->pending -= written;
+ len -= written;
}
+ }
+}
+
+static int alsa_run_out (HWVoiceOut *hw, int live)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ int decr;
+ snd_pcm_sframes_t avail;
- mixeng_clear (src, written);
- rpos = (rpos + written) % hw->samples;
- samples -= written;
+ avail = alsa_get_avail (alsa->handle);
+ if (avail < 0) {
+ dolog ("Could not get number of available playback frames\n");
+ return 0;
}
- exit:
- hw->rpos = rpos;
+ 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);
+ 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)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
- audfmt_e effective_fmt;
- int endianness;
- int err;
snd_pcm_t *handle;
- audsettings_t obt_as;
+ struct audsettings obt_as;
- 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)) {
return -1;
}
- err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
- if (err) {
- alsa_anal_close (&handle);
- return -1;
- }
-
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
- obt_as.fmt = effective_fmt;
+ obt_as.fmt = obt.fmt;
+ obt_as.endianness = obt.endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
- alsa->can_pause = obt.can_pause;
+ 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);
if (!alsa->pcm_buf) {
- dolog ("Could not allocate DAC buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
- alsa_anal_close (&handle);
+ dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
+ alsa_anal_close1 (&handle);
return -1;
}
alsa->handle = handle;
- alsa->was_enabled = 0;
return 0;
}
-static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+#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 (ctl == VOICE_CTL_PAUSE) {
+ err = snd_pcm_drop (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Could not stop %s\n", typ);
+ return -1;
+ }
+ }
+ else {
+ err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ 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, ...)
+{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
switch (cmd) {
case VOICE_ENABLE:
- ldebug ("enabling voice\n");
- audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples);
- if (alsa->can_pause) {
- /* Why this was_enabled madness is needed at all?? */
- if (alsa->was_enabled) {
- err = snd_pcm_pause (alsa->handle, 0);
- if (err < 0) {
- alsa_logerr (err, "Failed to resume playing\n");
- /* not fatal really */
- }
- }
- else {
- alsa->was_enabled = 1;
+ {
+ 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);
}
- break;
case VOICE_DISABLE:
ldebug ("disabling voice\n");
- if (alsa->can_pause) {
- err = snd_pcm_pause (alsa->handle, 1);
- if (err < 0) {
- alsa_logerr (err, "Failed to stop playing\n");
- /* not fatal really */
- }
+ if (hw->poll_mode) {
+ hw->poll_mode = 0;
+ alsa_fini_poll (&alsa->pollhlp);
}
- break;
+ return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
}
- return 0;
+
+ return -1;
}
-static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
+static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
- int endianness;
- int err;
- audfmt_e effective_fmt;
snd_pcm_t *handle;
- audsettings_t obt_as;
+ struct audsettings obt_as;
- 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)) {
return -1;
}
- err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
- if (err) {
- alsa_anal_close (&handle);
- return -1;
- }
-
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
- obt_as.fmt = effective_fmt;
+ obt_as.fmt = obt.fmt;
+ obt_as.endianness = obt.endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
- alsa->can_pause = obt.can_pause;
+ 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);
if (!alsa->pcm_buf) {
- dolog ("Could not allocate ADC buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
- alsa_anal_close (&handle);
+ dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
+ alsa_anal_close1 (&handle);
return -1;
}
{
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);
+ g_free (alsa->pcm_buf);
alsa->pcm_buf = NULL;
}
}
int i;
int live = audio_pcm_hw_get_live_in (hw);
int dead = hw->samples - live;
+ int decr;
struct {
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;
if (!dead) {
return 0;
}
- if (hw->wpos + dead > hw->samples) {
+ avail = alsa_get_avail (alsa->handle);
+ if (avail < 0) {
+ dolog ("Could not get number of captured frames\n");
+ return 0;
+ }
+
+ 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;
+ }
+ if (conf.verbose) {
+ dolog ("Resuming suspended input stream\n");
+ }
+ break;
+ default:
+ if (conf.verbose) {
+ dolog ("No frames available and ALSA state is %d\n", state);
+ }
+ return 0;
+ }
+ }
+
+ decr = audio_MIN (dead, avail);
+ if (!decr) {
+ return 0;
+ }
+
+ if (hw->wpos + decr > hw->samples) {
bufs[0].len = (hw->samples - hw->wpos);
- bufs[1].len = (dead - (hw->samples - hw->wpos));
+ bufs[1].len = (decr - (hw->samples - hw->wpos));
}
else {
- bufs[0].len = dead;
+ bufs[0].len = decr;
}
-
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;
while (len) {
nread = snd_pcm_readi (alsa->handle, src, len);
- if (nread < 0) {
+ if (nread <= 0) {
switch (nread) {
- case -EPIPE:
- if (!alsa_recover (alsa->handle)) {
- continue;
+ case 0:
+ if (conf.verbose) {
+ dolog ("Failed to read %ld frames (read zero)\n", len);
}
- dolog (
- "Failed to read %ld frames from %p, "
- "handle %p not prepared\n",
- len,
- src,
- alsa->handle
- );
goto exit;
- case -EAGAIN:
+ case -EPIPE:
+ if (alsa_recover (alsa->handle)) {
+ alsa_logerr (nread, "Failed to read %ld frames\n", len);
+ goto exit;
+ }
+ if (conf.verbose) {
+ dolog ("Recovering from capture xrun\n");
+ }
continue;
+ case -EAGAIN:
+ goto exit;
+
default:
alsa_logerr (
nread,
}
}
- 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, ...)
{
- (void) hw;
- (void) cmd;
- return 0;
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ {
+ 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");
+ 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 void *alsa_audio_init (void)
}
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", &conf.period_size_out_overriden, 0},
- {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
- "DAC buffer size", &conf.buffer_size_out_overriden, 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", &conf.period_size_in_overriden, 0},
- {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
- "ADC buffer size", &conf.buffer_size_in_overriden, 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},
- {NULL, 0, NULL, NULL, NULL, 0}
+ {
+ .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 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
+ .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 / qemu.git / blobdiff