* 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 "qemu/module.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;
+ AudioState *s;
};
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 {
HWVoiceIn hw;
snd_pcm_t *handle;
- void *pcm_buf;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceIn;
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;
break;
case SND_PCM_STATE_PREPARED:
- audio_run ("alsa run (prepared)");
+ audio_run(hlp->s, "alsa run (prepared)");
break;
case SND_PCM_STATE_RUNNING:
- audio_run ("alsa run (running)");
+ audio_run(hlp->s, "alsa run (running)");
break;
default:
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, int endianness)
+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:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
return SND_PCM_FORMAT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
}
return SND_PCM_FORMAT_U16_LE;
}
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
}
return SND_PCM_FORMAT_S32_LE;
}
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
}
return SND_PCM_FORMAT_U32_LE;
}
+ case AUDIO_FORMAT_F32:
+ if (endianness) {
+ return SND_PCM_FORMAT_FLOAT_BE;
+ } else {
+ return SND_PCM_FORMAT_FLOAT_LE;
+ }
+
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
}
}
-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;
+
+ case SND_PCM_FORMAT_FLOAT_LE:
+ *endianness = 0;
+ *fmt = AUDIO_FORMAT_F32;
+ break;
+
+ case SND_PCM_FORMAT_FLOAT_BE:
+ *endianness = 1;
+ *fmt = AUDIO_FORMAT_F32;
break;
default:
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt,
- snd_pcm_format_t obtfmt)
+ snd_pcm_format_t obtfmt,
+ AudiodevAlsaPerDirectionOptions *apdo)
{
- 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 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,
- ALSAConf *conf)
+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);
goto err;
}
- if (nchannels != 1 && nchannels != 2) {
- alsa_logerr2 (err, typ,
- "Can not handle obtained number of channels %d\n",
- nchannels);
- 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 (apdo->period_length) {
+ int dir = 0;
+ unsigned int ptime = apdo->period_length;
- 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,
- &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;
obt->nchannels != req->nchannels ||
obt->freq != req->freq) {
dolog ("Audio parameters for %s\n", typ);
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info(req, obt, obtfmt, apdo);
}
#ifdef DEBUG
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info(req, obt, obtfmt, pdo);
#endif
return 0;
return -1;
}
-static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
+static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
- snd_pcm_sframes_t avail;
-
- avail = snd_pcm_avail_update (handle);
- if (avail < 0) {
- if (avail == -EPIPE) {
- if (!alsa_recover (handle)) {
- avail = snd_pcm_avail_update (handle);
- }
- }
-
- if (avail < 0) {
- alsa_logerr (avail,
- "Could not obtain number of available frames\n");
- return -1;
- }
- }
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ size_t pos = 0;
+ size_t len_frames = len / hw->info.bytes_per_frame;
+
+ while (len_frames) {
+ char *src = advance(buf, pos);
+ snd_pcm_sframes_t written;
+
+ written = snd_pcm_writei(alsa->handle, src, len_frames);
+
+ if (written <= 0) {
+ switch (written) {
+ case 0:
+ trace_alsa_wrote_zero(len_frames);
+ return pos;
+
+ case -EPIPE:
+ if (alsa_recover(alsa->handle)) {
+ alsa_logerr(written, "Failed to write %zu frames\n",
+ len_frames);
+ return pos;
+ }
+ 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 %zu frames\n",
+ len_frames);
+ return pos;
+ }
+ trace_alsa_resume_out();
+ continue;
- return avail;
-}
+ case -EAGAIN:
+ return pos;
-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:
- trace_alsa_wrote_zero(len);
- return;
-
- case -EPIPE:
- if (alsa_recover (alsa->handle)) {
- alsa_logerr (written, "Failed to write %d frames\n",
- len);
- return;
- }
- 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:
- return;
-
- default:
- alsa_logerr (written, "Failed to write %d frames from %p\n",
- len, src);
- return;
- }
+ default:
+ alsa_logerr(written, "Failed to write %zu frames from %p\n",
+ len, src);
+ return pos;
}
-
- 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;
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of available playback frames\n");
- return 0;
+ pos += written * hw->info.bytes_per_frame;
+ if (written < len_frames) {
+ break;
+ }
+ len_frames -= written;
}
- 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;
+ return pos;
}
static void alsa_fini_out (HWVoiceOut *hw)
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
-
- g_free(alsa->pcm_buf);
- alsa->pcm_buf = NULL;
}
static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
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, conf)) {
+
+ 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);
- if (!alsa->pcm_buf) {
- 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->pollhlp.s = hw->s;
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
return 0;
}
-static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void alsa_enable_out(HWVoiceOut *hw, bool enable)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- va_list ap;
- int poll_mode;
+ if (enable) {
+ bool poll_mode = apdo->try_poll;
- 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);
+ ldebug("enabling voice\n");
+ if (poll_mode && alsa_poll_out(hw)) {
+ poll_mode = 0;
}
-
- case VOICE_DISABLE:
- ldebug ("disabling voice\n");
+ hw->poll_mode = poll_mode;
+ alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
+ } else {
+ ldebug("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
- alsa_fini_poll (&alsa->pollhlp);
+ alsa_fini_poll(&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
+ alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
}
-
- return -1;
}
static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
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, conf)) {
+
+ 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);
- if (!alsa->pcm_buf) {
- dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- alsa_anal_close1 (&handle);
- return -1;
- }
-
+ alsa->pollhlp.s = hw->s;
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
-
- g_free(alsa->pcm_buf);
- alsa->pcm_buf = NULL;
}
-static int alsa_run_in (HWVoiceIn *hw)
+static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
- int hwshift = hw->info.shift;
- 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] = {
- { .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;
- }
-
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of captured frames\n");
- return 0;
- }
+ size_t pos = 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;
- }
- trace_alsa_resume_in();
- break;
- default:
- trace_alsa_no_frames(state);
- return 0;
- }
- }
+ while (len) {
+ void *dst = advance(buf, pos);
+ snd_pcm_sframes_t nread;
- decr = audio_MIN (dead, avail);
- if (!decr) {
- return 0;
- }
+ nread = snd_pcm_readi(
+ alsa->handle, dst, len / hw->info.bytes_per_frame);
- if (hw->wpos + decr > hw->samples) {
- bufs[0].len = (hw->samples - hw->wpos);
- bufs[1].len = (decr - (hw->samples - hw->wpos));
- }
- else {
- bufs[0].len = decr;
- }
+ if (nread <= 0) {
+ switch (nread) {
+ case 0:
+ trace_alsa_read_zero(len);
+ return pos;;
- for (i = 0; i < 2; ++i) {
- void *src;
- struct st_sample *dst;
- snd_pcm_sframes_t nread;
- snd_pcm_uframes_t len;
-
- len = bufs[i].len;
-
- src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
- dst = hw->conv_buf + bufs[i].add;
-
- while (len) {
- nread = snd_pcm_readi (alsa->handle, src, len);
-
- if (nread <= 0) {
- switch (nread) {
- case 0:
- trace_alsa_read_zero(len);
- goto exit;
-
- case -EPIPE:
- if (alsa_recover (alsa->handle)) {
- alsa_logerr (nread, "Failed to read %ld frames\n", len);
- goto exit;
- }
- trace_alsa_xrun_in();
- continue;
-
- case -EAGAIN:
- goto exit;
-
- default:
- alsa_logerr (
- nread,
- "Failed to read %ld frames from %p\n",
- len,
- src
- );
- goto exit;
+ case -EPIPE:
+ if (alsa_recover(alsa->handle)) {
+ alsa_logerr(nread, "Failed to read %zu frames\n", len);
+ return pos;
}
- }
+ trace_alsa_xrun_in();
+ continue;
- hw->conv (dst, src, nread);
+ case -EAGAIN:
+ return pos;
- src = advance (src, nread << hwshift);
- dst += nread;
-
- read_samples += nread;
- len -= nread;
+ default:
+ alsa_logerr(nread, "Failed to read %zu frames to %p\n",
+ len, dst);
+ return pos;;
+ }
}
- }
- exit:
- hw->wpos = (hw->wpos + read_samples) % hw->samples;
- return read_samples;
-}
+ pos += nread * hw->info.bytes_per_frame;
+ len -= nread * hw->info.bytes_per_frame;
+ }
-static int alsa_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
+ return pos;
}
-static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void alsa_enable_in(HWVoiceIn *hw, bool enable)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- va_list ap;
- int poll_mode;
+ if (enable) {
+ bool poll_mode = apdo->try_poll;
- 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);
+ ldebug("enabling voice\n");
+ if (poll_mode && alsa_poll_in(hw)) {
+ poll_mode = 0;
}
+ hw->poll_mode = poll_mode;
- case VOICE_DISABLE:
+ alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
+ } else {
ldebug ("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
- alsa_fini_poll (&alsa->pollhlp);
+ alsa_fini_poll(&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
+ 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_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
+{
+ if (!apdo->has_try_poll) {
+ apdo->try_poll = true;
+ apdo->has_try_poll = true;
+ }
+}
-static void *alsa_audio_init (void)
+static void *alsa_audio_init(Audiodev *dev)
{
- ALSAConf *conf = g_malloc(sizeof(ALSAConf));
- *conf = glob_conf;
- return conf;
+ 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 void alsa_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option alsa_options[] = {
- {
- .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 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,
+ .run_buffer_out = audio_generic_run_buffer_out,
+ .enable_out = alsa_enable_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
- .run_in = alsa_run_in,
.read = alsa_read,
- .ctl_in = alsa_ctl_in,
+ .enable_in = alsa_enable_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 / qemu.git / blobdiff