qemu/audio/alsaaudio.c
Michael Walle 00e076795f audio: split sample conversion and volume mixing
Refactor the volume mixing, so it can be reused for capturing devices.
Additionally, it removes superfluous multiplications with the nominal
volume within the hardware voice code path.

Signed-off-by: Michael Walle <michael@walle.cc>
Signed-off-by: malc <av1474@comtv.ru>
2011-01-12 18:36:22 +03:00

1261 lines
32 KiB
C

/*
* 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-char.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);
}
qemu_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");
qemu_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);
}
qemu_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) {
qemu_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) {
qemu_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)
};