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qemu/audio/alsaaudio.c

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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 <alsa/asoundlib.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;
struct pollhlp pollhlp;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
void *pcm_buf;
struct pollhlp pollhlp;
} 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 = 4096,
.period_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;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
};
static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}
static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
int err,
const char *typ,
const char *fmt,
...
)
{
va_list ap;
AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}
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\n", *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\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 snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
{
switch (fmt) {
case AUD_FMT_S8:
return SND_PCM_FORMAT_S8;
case AUD_FMT_U8:
return SND_PCM_FORMAT_U8;
case AUD_FMT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
else {
return SND_PCM_FORMAT_S16_LE;
}
case AUD_FMT_U16:
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);
#ifdef DEBUG_AUDIO
abort ();
#endif
return SND_PCM_FORMAT_U8;
}
}
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
*fmt = AUD_FMT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
*fmt = AUD_FMT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
*fmt = AUD_FMT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
*fmt = AUD_FMT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
*fmt = AUD_FMT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
*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;
}
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt,
snd_pcm_format_t obtfmt)
{
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);
}
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) {
dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to get current software parameters\n");
return;
}
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
if (err < 0) {
dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software threshold to %ld\n",
threshold);
return;
}
err = snd_pcm_sw_params (handle, sw_params);
if (err < 0) {
dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software parameters\n");
return;
}
}
static int alsa_open (int in, struct alsa_params_req *req,
struct alsa_params_obt *obt, snd_pcm_t **handlep)
{
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;
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_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':\n", 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\n");
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\n");
goto err;
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
if (err < 0 && conf.verbose) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", 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\n", 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\n",
req->nchannels);
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 (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
);
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);
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 (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,
&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;
}
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);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
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\n");
goto err;
}
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to get format\n");
goto err;
}
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);
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->nchannels = nchannels;
obt->freq = freq;
obt->samples = obt_buffer_size;
*handlep = handle;
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);
}
#ifdef DEBUG
alsa_dump_info (req, obt, obtfmt);
#endif
return 0;
err:
alsa_anal_close1 (&handle);
return -1;
}
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
{
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;
}
}
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;
}
}
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;
}
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;
}
static void alsa_fini_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
if (alsa->pcm_buf) {
g_free (alsa->pcm_buf);
alsa->pcm_buf = NULL;
}
}
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;
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;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
obt_as.endianness = obt.endianness;
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->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\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:
{
va_list ap;
int poll_mode;
va_start (ap, cmd);
poll_mode = va_arg (ap, int);
va_end (ap);
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_out (hw)) {
poll_mode = 0;
}
hw->poll_mode = poll_mode;
return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
}
case VOICE_DISABLE:
ldebug ("disabling voice\n");
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)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
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;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
obt_as.endianness = obt.endianness;
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->handle = handle;
return 0;
}
static void alsa_fini_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
if (alsa->pcm_buf) {
g_free (alsa->pcm_buf);
alsa->pcm_buf = NULL;
}
}
static int alsa_run_in (HWVoiceIn *hw)
{
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;
}
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 = (decr - (hw->samples - hw->wpos));
}
else {
bufs[0].len = decr;
}
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:
if (conf.verbose) {
dolog ("Failed to read %ld frames (read zero)\n", len);
}
goto exit;
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,
"Failed to read %ld frames from %p\n",
len,
src
);
goto exit;
}
}
hw->conv (dst, src, nread);
src = advance (src, nread << hwshift);
dst += nread;
read_samples += nread;
len -= nread;
}
}
exit:
hw->wpos = (hw->wpos + read_samples) % hw->samples;
return read_samples;
}
static int alsa_read (SWVoiceIn *sw, void *buf, int size)
{
return audio_pcm_sw_read (sw, buf, size);
}
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
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)
{
return &conf;
}
static void alsa_audio_fini (void *opaque)
{
(void) opaque;
}
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 struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.run_out = alsa_run_out,
.write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.run_in = alsa_run_in,
.read = alsa_read,
.ctl_in = alsa_ctl_in,
};
struct audio_driver alsa_audio_driver = {
.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)
};
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