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qemu/audio/alsaaudio.c
Marc-André Lureau 097409f57d audio: remove AudioMixengBackendClass.name 
Simply use the class name instead.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
2026-02-23 14:38:22 +01:00

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/*
* QEMU ALSA audio driver
*
* Copyright (c) 2005 Vassili Karpov (malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* 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/main-loop.h"
#include "qemu/module.h"
#include "qemu/error-report.h"
#include "qemu/audio.h"
#include "qom/object.h"
#include "trace.h"
#pragma GCC diagnostic ignored "-Waddress"
#include "audio_int.h"
#define TYPE_AUDIO_ALSA "audio-alsa"
OBJECT_DECLARE_SIMPLE_TYPE(AudioALSA, AUDIO_ALSA)
static AudioBackendClass *audio_alsa_parent_class;
struct AudioALSA {
AudioMixengBackend parent_obj;
};
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
int count;
int mask;
AudioMixengBackend *s;
};
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
snd_pcm_t *handle;
struct pollhlp pollhlp;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
struct pollhlp pollhlp;
} ALSAVoiceIn;
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
int nchannels;
};
struct alsa_params_obt {
int freq;
AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
};
static void G_GNUC_PRINTF(2, 3) alsa_logerr(int err, const char *fmt, ...)
{
va_list ap;
error_printf("alsa: ");
va_start(ap, fmt);
error_vprintf(fmt, ap);
va_end(ap);
error_printf(" Reason: %s", snd_strerror(err));
error_printf("\n");
}
static void G_GNUC_PRINTF(3, 4) alsa_logerr2(int err, const char *typ,
const char *fmt, ...)
{
va_list ap;
error_printf("alsa: Could not initialize %s:", typ);
va_start(ap, fmt);
error_vprintf(fmt, ap);
va_end(ap);
error_printf(" Reason: %s", snd_strerror(err));
error_printf("\n");
}
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) {
alsa_logerr(err, "Failed to close PCM handle %p", *handlep);
}
*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", 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", 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) {
warn_report("alsa_poll_handler: poll %s", 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(hlp->s, "alsa run (prepared)");
break;
case SND_PCM_STATE_RUNNING:
audio_run(hlp->s, "alsa run (running)");
break;
default:
warn_report("alsa: Unexpected state %d", 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) {
warn_report("alsa: Could not initialize poll mode: "
"Invalid number of poll descriptors %d", count);
return -1;
}
pfds = g_new0(struct pollfd, count);
err = snd_pcm_poll_descriptors (handle, pfds, count);
if (err < 0) {
alsa_logerr(err, "Could not initialize poll mode: Could not obtain poll descriptors");
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 snd_pcm_format_t aud_to_alsafmt(AudioFormat fmt, bool big_endian)
{
switch (fmt) {
case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
case AUDIO_FORMAT_S16:
return big_endian ? SND_PCM_FORMAT_S16_BE : SND_PCM_FORMAT_S16_LE;
case AUDIO_FORMAT_U16:
return big_endian ? SND_PCM_FORMAT_U16_BE : SND_PCM_FORMAT_U16_LE;
case AUDIO_FORMAT_S32:
return big_endian ? SND_PCM_FORMAT_S32_BE : SND_PCM_FORMAT_S32_LE;
case AUDIO_FORMAT_U32:
return big_endian ? SND_PCM_FORMAT_U32_BE : SND_PCM_FORMAT_U32_LE;
case AUDIO_FORMAT_F32:
return big_endian ? SND_PCM_FORMAT_FLOAT_BE : SND_PCM_FORMAT_FLOAT_LE;
default:
warn_report("alsa: Internal logic error: Bad audio format %d", fmt);
return SND_PCM_FORMAT_U8;
}
}
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
*fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
*fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
*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:
warn_report("alsa: Unrecognized audio format %d", alsafmt);
return -1;
}
return 0;
}
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
int err;
snd_pcm_sw_params_t *sw_params;
snd_pcm_sw_params_alloca (&sw_params);
err = snd_pcm_sw_params_current (handle, sw_params);
if (err < 0) {
error_report("alsa: Could not fully initialize DAC");
alsa_logerr(err, "Failed to get current software parameters");
return;
}
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
if (err < 0) {
error_report("alsa: Could not fully initialize DAC");
alsa_logerr(err, "Failed to set software threshold to %ld", threshold);
return;
}
err = snd_pcm_sw_params (handle, sw_params);
if (err < 0) {
error_report("alsa: Could not fully initialize DAC");
alsa_logerr(err, "Failed to set software parameters");
return;
}
}
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;
unsigned int freq, nchannels;
const char *pcm_name = 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;
snd_pcm_hw_params_alloca (&hw_params);
err = snd_pcm_open (
&handle,
pcm_name,
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK
);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to open `%s'", pcm_name);
return -1;
}
err = snd_pcm_hw_params_any (handle, hw_params);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to initialize hardware parameters");
goto err;
}
err = snd_pcm_hw_params_set_access (
handle,
hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED
);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set access type");
goto err;
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set format %d", req->fmt);
}
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set frequency %d", req->freq);
goto err;
}
err = snd_pcm_hw_params_set_channels_near (
handle,
hw_params,
&nchannels
);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set number of channels %d", req->nchannels);
goto err;
}
if (apdo->buffer_length) {
int dir = 0;
unsigned int btime = apdo->buffer_length;
err = snd_pcm_hw_params_set_buffer_time_near(
handle, hw_params, &btime, &dir);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32,
apdo->buffer_length);
goto err;
}
if (apdo->has_buffer_length && btime != apdo->buffer_length) {
warn_report("alsa: Requested buffer time %" PRId32 " was rejected, using %u",
apdo->buffer_length, btime);
}
}
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);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set period time to %" PRId32,
apdo->period_length);
goto err;
}
if (apdo->has_period_length && ptime != apdo->period_length) {
warn_report("alsa: Requested period time %" PRId32 " was rejected, using %d",
apdo->period_length, ptime);
}
}
err = snd_pcm_hw_params (handle, hw_params);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to apply audio parameters");
goto err;
}
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to get buffer size");
goto err;
}
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to get format");
goto err;
}
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
error_report("alsa: Invalid format was returned %d", obtfmt);
goto err;
}
err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr2(err, typ, "Could not prepare handle %p", handle);
goto err;
}
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->freq = freq;
obt->samples = obt_buffer_size;
*handlep = handle;
trace_alsa_info_params(req->fmt, obtfmt, req->nchannels, obt->nchannels,
req->freq, obt->freq);
trace_alsa_info_samples(apdo->buffer_length, apdo->period_length, obt->samples);
return 0;
err:
alsa_anal_close1 (&handle);
return -1;
}
static size_t alsa_buffer_get_free(HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
snd_pcm_sframes_t avail;
size_t alsa_free, generic_free, generic_in_use;
avail = snd_pcm_avail_update(alsa->handle);
if (avail < 0) {
if (avail == -EPIPE) {
if (!alsa_recover(alsa->handle)) {
avail = snd_pcm_avail_update(alsa->handle);
}
}
if (avail < 0) {
alsa_logerr(avail, "Could not obtain number of available frames");
avail = 0;
}
}
alsa_free = avail * hw->info.bytes_per_frame;
generic_free = audio_generic_buffer_get_free(hw);
generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
if (generic_in_use) {
/*
* This code can only be reached in the unlikely case that
* snd_pcm_avail_update() returned a larger number of frames
* than snd_pcm_writei() could write. Make sure that all
* remaining bytes in the generic buffer can be written.
*/
alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
}
return alsa_free;
}
static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
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", 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", len_frames);
return pos;
}
trace_alsa_resume_out();
continue;
case -EAGAIN:
return pos;
default:
alsa_logerr(written, "Failed to write %zu frames from %p", len_frames, src);
return pos;
}
}
pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
len_frames -= written;
}
return pos;
}
static void alsa_fini_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
trace_alsa_fini_out();
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
}
static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
Audiodev *dev = hw->s->dev;
req.fmt = aud_to_alsafmt (as->fmt, as->big_endian);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (alsa_open(0, &req, &obt, &handle, dev)) {
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
obt_as.big_endian = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
return 0;
}
#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", typ);
return -1;
}
} else {
err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr(err, "Could not prepare handle for %s", 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", typ);
return -1;
}
}
}
return 0;
}
static void alsa_enable_out(HWVoiceOut *hw, bool enable)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
AudiodevAlsaPerDirectionOptions *apdo = hw->s->dev->u.alsa.out;
trace_alsa_enable_out(enable);
if (enable) {
bool poll_mode = apdo->try_poll;
if (poll_mode && alsa_poll_out(hw)) {
poll_mode = 0;
}
hw->poll_mode = poll_mode;
alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
} else {
if (hw->poll_mode) {
hw->poll_mode = 0;
alsa_fini_poll(&alsa->pollhlp);
}
alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
}
}
static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
Audiodev *dev = hw->s->dev;
req.fmt = aud_to_alsafmt (as->fmt, as->big_endian);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (alsa_open(1, &req, &obt, &handle, dev)) {
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
obt_as.big_endian = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
return 0;
}
static void alsa_fini_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
}
static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
size_t pos = 0;
while (len) {
void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
nread = snd_pcm_readi(
alsa->handle, dst, len / hw->info.bytes_per_frame);
if (nread <= 0) {
switch (nread) {
case 0:
trace_alsa_read_zero(len);
return pos;
case -EPIPE:
if (alsa_recover(alsa->handle)) {
alsa_logerr(nread, "Failed to read %zu frames", len);
return pos;
}
trace_alsa_xrun_in();
continue;
case -EAGAIN:
return pos;
default:
alsa_logerr(nread, "Failed to read %zu frames to %p", len, dst);
return pos;
}
}
pos += nread * hw->info.bytes_per_frame;
len -= nread * hw->info.bytes_per_frame;
}
return pos;
}
static void alsa_enable_in(HWVoiceIn *hw, bool enable)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
AudiodevAlsaPerDirectionOptions *apdo = hw->s->dev->u.alsa.in;
trace_alsa_enable_in(enable);
if (enable) {
bool poll_mode = apdo->try_poll;
if (poll_mode && alsa_poll_in(hw)) {
poll_mode = 0;
}
hw->poll_mode = poll_mode;
alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
} else {
if (hw->poll_mode) {
hw->poll_mode = 0;
alsa_fini_poll(&alsa->pollhlp);
}
alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
}
}
static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
{
if (!apdo->has_try_poll) {
apdo->try_poll = false;
apdo->has_try_poll = true;
}
}
static bool
audio_alsa_realize(AudioBackend *abe, Audiodev *dev, Error **errp)
{
AudiodevAlsaOptions *aopts;
assert(dev->driver == AUDIODEV_DRIVER_ALSA);
aopts = &dev->u.alsa;
alsa_init_per_direction(aopts->in);
alsa_init_per_direction(aopts->out);
/* don't set has_* so alsa_open can identify it wasn't set by the user */
if (!dev->u.alsa.out->has_period_length) {
/* 256 frames assuming 44100Hz */
dev->u.alsa.out->period_length = 5805;
}
if (!dev->u.alsa.out->has_buffer_length) {
/* 4096 frames assuming 44100Hz */
dev->u.alsa.out->buffer_length = 92880;
}
if (!dev->u.alsa.in->has_period_length) {
/* 256 frames assuming 44100Hz */
dev->u.alsa.in->period_length = 5805;
}
if (!dev->u.alsa.in->has_buffer_length) {
/* 4096 frames assuming 44100Hz */
dev->u.alsa.in->buffer_length = 92880;
}
return audio_alsa_parent_class->realize(abe, dev, errp);
}
static void audio_alsa_class_init(ObjectClass *klass, const void *data)
{
AudioBackendClass *b = AUDIO_BACKEND_CLASS(klass);
AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_CLASS(klass);
audio_alsa_parent_class = AUDIO_BACKEND_CLASS(object_class_get_parent(klass));
b->realize = audio_alsa_realize;
k->max_voices_out = INT_MAX;
k->max_voices_in = INT_MAX;
k->voice_size_out = sizeof(ALSAVoiceOut);
k->voice_size_in = sizeof(ALSAVoiceIn);
k->init_out = alsa_init_out;
k->fini_out = alsa_fini_out;
k->write = alsa_write;
k->buffer_get_free = alsa_buffer_get_free;
k->run_buffer_out = audio_generic_run_buffer_out;
k->enable_out = alsa_enable_out;
k->init_in = alsa_init_in;
k->fini_in = alsa_fini_in;
k->read = alsa_read;
k->run_buffer_in = audio_generic_run_buffer_in;
k->enable_in = alsa_enable_in;
}
static const TypeInfo audio_types[] = {
{
.name = TYPE_AUDIO_ALSA,
.parent = TYPE_AUDIO_MIXENG_BACKEND,
.instance_size = sizeof(AudioALSA),
.class_init = audio_alsa_class_init,
},
};
DEFINE_TYPES(audio_types)
module_obj(TYPE_AUDIO_ALSA);
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