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qemu-e2k/audio/alsaaudio.c
ths fe8f096b16 Spelling fix, by Stuart Brady. 
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@3076 c046a42c-6fe2-441c-8c8c-71466251a162
2007-07-12 10:59:21 +00:00

1001 lines
26 KiB
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 "vl.h"
#define AUDIO_CAP "alsa"
#include "audio_int.h"
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
void *pcm_buf;
snd_pcm_t *handle;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
void *pcm_buf;
} 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 = {
#define DEFAULT_BUFFER_SIZE 1024
#define DEFAULT_PERIOD_SIZE 256
#ifdef HIGH_LATENCY
.size_in_usec_in = 1,
.size_in_usec_out = 1,
#endif
.pcm_name_out = "default",
.pcm_name_in = "default",
#ifdef HIGH_LATENCY
.buffer_size_in = 400000,
.period_size_in = 400000 / 4,
.buffer_size_out = 400000,
.period_size_out = 400000 / 4,
#else
.buffer_size_in = DEFAULT_BUFFER_SIZE * 4,
.period_size_in = DEFAULT_PERIOD_SIZE * 4,
.buffer_size_out = DEFAULT_BUFFER_SIZE,
.period_size_out = DEFAULT_PERIOD_SIZE,
.buffer_size_in_overridden = 0,
.buffer_size_out_overridden = 0,
.period_size_in_overridden = 0,
.period_size_out_overridden = 0,
#endif
.threshold = 0,
.verbose = 0
};
struct alsa_params_req {
int freq;
audfmt_e fmt;
int nchannels;
unsigned int buffer_size;
unsigned int period_size;
};
struct alsa_params_obt {
int freq;
audfmt_e fmt;
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_anal_close (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 int alsa_write (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}
static int aud_to_alsafmt (audfmt_e fmt)
{
switch (fmt) {
case AUD_FMT_S8:
return SND_PCM_FORMAT_S8;
case AUD_FMT_U8:
return SND_PCM_FORMAT_U8;
case AUD_FMT_S16:
return SND_PCM_FORMAT_S16_LE;
case AUD_FMT_U16:
return SND_PCM_FORMAT_U16_LE;
case AUD_FMT_S32:
return SND_PCM_FORMAT_S32_LE;
case AUD_FMT_U32:
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 (int 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;
}
#if defined DEBUG_MISMATCHES || defined DEBUG
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt)
{
dolog ("parameter | requested value | obtained value\n");
dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
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);
}
#endif
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, freq, nchannels;
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
unsigned int period_size, buffer_size;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
freq = req->freq;
period_size = req->period_size;
buffer_size = req->buffer_size;
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':\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) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
goto err;
}
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
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 (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
if (!buffer_size) {
buffer_size = DEFAULT_BUFFER_SIZE;
period_size= DEFAULT_PERIOD_SIZE;
}
}
if (buffer_size) {
if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
if (period_size) {
err = snd_pcm_hw_params_set_period_time_near (
handle,
hw_params,
&period_size,
0
);
if (err < 0) {
alsa_logerr2 (err, typ,
"Failed to set period time %d\n",
req->period_size);
goto err;
}
}
err = snd_pcm_hw_params_set_buffer_time_near (
handle,
hw_params,
&buffer_size,
0
);
if (err < 0) {
alsa_logerr2 (err, typ,
"Failed to set buffer time %d\n",
req->buffer_size);
goto err;
}
}
else {
int dir;
snd_pcm_uframes_t minval;
if (period_size) {
minval = period_size;
dir = 0;
err = snd_pcm_hw_params_get_period_size_min (
hw_params,
&minval,
&dir
);
if (err < 0) {
alsa_logerr (
err,
"Could not get minmal period size for %s\n",
typ
);
}
else {
if (period_size < minval) {
if ((in && conf.period_size_in_overridden)
|| (!in && conf.period_size_out_overridden)) {
dolog ("%s period size(%d) is less "
"than minmal period size(%ld)\n",
typ,
period_size,
minval);
}
period_size = minval;
}
}
err = snd_pcm_hw_params_set_period_size (
handle,
hw_params,
period_size,
0
);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set period size %d\n",
req->period_size);
goto err;
}
}
minval = buffer_size;
err = snd_pcm_hw_params_get_buffer_size_min (
hw_params,
&minval
);
if (err < 0) {
alsa_logerr (err, "Could not get minmal buffer size for %s\n",
typ);
}
else {
if (buffer_size < minval) {
if ((in && conf.buffer_size_in_overridden)
|| (!in && conf.buffer_size_out_overridden)) {
dolog (
"%s buffer size(%d) is less "
"than minimal buffer size(%ld)\n",
typ,
buffer_size,
minval
);
}
buffer_size = minval;
}
}
err = snd_pcm_hw_params_set_buffer_size (
handle,
hw_params,
buffer_size
);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
req->buffer_size);
goto err;
}
}
}
else {
dolog ("warning: Buffer size is not set\n");
}
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_prepare (handle);
if (err < 0) {
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
goto err;
}
if (!in && conf.threshold) {
snd_pcm_uframes_t threshold;
int bytes_per_sec;
bytes_per_sec = freq
<< (nchannels == 2)
<< (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
threshold = (conf.threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
obt->fmt = req->fmt;
obt->nchannels = nchannels;
obt->freq = freq;
obt->samples = obt_buffer_size;
*handlep = handle;
#if defined DEBUG_MISMATCHES || defined DEBUG
if (obt->fmt != req->fmt ||
obt->nchannels != req->nchannels ||
obt->freq != req->freq) {
dolog ("Audio paramters mismatch for %s\n", typ);
alsa_dump_info (req, obt);
}
#endif
#ifdef DEBUG
alsa_dump_info (req, obt);
#endif
return 0;
err:
alsa_anal_close (&handle);
return -1;
}
static int alsa_recover (snd_pcm_t *handle)
{
int err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
return -1;
}
return 0;
}
static 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 int alsa_run_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
int rpos, live, decr;
int samples;
uint8_t *dst;
st_sample_t *src;
snd_pcm_sframes_t avail;
live = audio_pcm_hw_get_live_out (hw);
if (!live) {
return 0;
}
avail = 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);
samples = decr;
rpos = hw->rpos;
while (samples) {
int left_till_end_samples = hw->samples - rpos;
int len = audio_MIN (samples, left_till_end_samples);
snd_pcm_sframes_t written;
src = hw->mix_buf + rpos;
dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
hw->clip (dst, src, len);
while (len) {
written = snd_pcm_writei (alsa->handle, dst, len);
if (written <= 0) {
switch (written) {
case 0:
if (conf.verbose) {
dolog ("Failed to write %d frames (wrote zero)\n", len);
}
goto exit;
case -EPIPE:
if (alsa_recover (alsa->handle)) {
alsa_logerr (written, "Failed to write %d frames\n",
len);
goto exit;
}
if (conf.verbose) {
dolog ("Recovering from playback xrun\n");
}
continue;
case -EAGAIN:
goto exit;
default:
alsa_logerr (written, "Failed to write %d frames to %p\n",
len, dst);
goto exit;
}
}
rpos = (rpos + written) % hw->samples;
samples -= written;
len -= written;
dst = advance (dst, written << hw->info.shift);
src += written;
}
}
exit:
hw->rpos = rpos;
return decr;
}
static void alsa_fini_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle);
if (alsa->pcm_buf) {
qemu_free (alsa->pcm_buf);
alsa->pcm_buf = NULL;
}
}
static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
audfmt_e effective_fmt;
int endianness;
int err;
snd_pcm_t *handle;
audsettings_t obt_as;
req.fmt = aud_to_alsafmt (as->fmt);
req.freq = as->freq;
req.nchannels = as->nchannels;
req.period_size = conf.period_size_out;
req.buffer_size = conf.buffer_size_out;
if (alsa_open (0, &req, &obt, &handle)) {
return -1;
}
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
if (err) {
alsa_anal_close (&handle);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
obt_as.endianness = 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_close (&handle);
return -1;
}
alsa->handle = handle;
return 0;
}
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
{
int err;
if (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;
}
}
return 0;
}
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
switch (cmd) {
case VOICE_ENABLE:
ldebug ("enabling voice\n");
return alsa_voice_ctl (alsa->handle, "playback", 0);
case VOICE_DISABLE:
ldebug ("disabling voice\n");
return alsa_voice_ctl (alsa->handle, "playback", 1);
}
return -1;
}
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
int endianness;
int err;
audfmt_e effective_fmt;
snd_pcm_t *handle;
audsettings_t obt_as;
req.fmt = aud_to_alsafmt (as->fmt);
req.freq = as->freq;
req.nchannels = as->nchannels;
req.period_size = conf.period_size_in;
req.buffer_size = conf.buffer_size_in;
if (alsa_open (1, &req, &obt, &handle)) {
return -1;
}
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
if (err) {
alsa_anal_close (&handle);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
obt_as.endianness = 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_close (&handle);
return -1;
}
alsa->handle = handle;
return 0;
}
static void alsa_fini_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle);
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] = {
{ hw->wpos, 0 },
{ 0, 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 (alsa->handle) == SND_PCM_STATE_PREPARED)) {
avail = hw->samples;
}
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;
st_sample_t *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, &nominal_volume);
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:
ldebug ("enabling voice\n");
return alsa_voice_ctl (alsa->handle, "capture", 0);
case VOICE_DISABLE:
ldebug ("disabling voice\n");
return alsa_voice_ctl (alsa->handle, "capture", 1);
}
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[] = {
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
"DAC period size", &conf.period_size_out_overridden, 0},
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
"DAC buffer size", &conf.buffer_size_out_overridden, 0},
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
"ADC period size", &conf.period_size_in_overridden, 0},
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
"ADC buffer size", &conf.buffer_size_in_overridden, 0},
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
"(undocumented)", NULL, 0},
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
"DAC device name (for instance dmix)", NULL, 0},
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
"ADC device name", NULL, 0},
{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
"Behave in a more verbose way", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
};
static struct audio_pcm_ops alsa_pcm_ops = {
alsa_init_out,
alsa_fini_out,
alsa_run_out,
alsa_write,
alsa_ctl_out,
alsa_init_in,
alsa_fini_in,
alsa_run_in,
alsa_read,
alsa_ctl_in
};
struct audio_driver alsa_audio_driver = {
INIT_FIELD (name = ) "alsa",
INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
INIT_FIELD (options = ) alsa_options,
INIT_FIELD (init = ) alsa_audio_init,
INIT_FIELD (fini = ) alsa_audio_fini,
INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
INIT_FIELD (can_be_default = ) 1,
INIT_FIELD (max_voices_out = ) INT_MAX,
INIT_FIELD (max_voices_in = ) INT_MAX,
INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
};
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