/**
* @file alsa-dsp.c
* @brief Alsa External plugin: I/O plugin
* <p>
* Copyright (C) 2006 Nokia Corporation
* <p>
* Contact: Eduardo Bezerra Valentin <eduardo.valentin@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
* */
#include <stdio.h>
#include <sys/ioctl.h>
#include <alsa/asoundlib.h>
#include <alsa/pcm_external.h>
#include "list.h"
#include "debug.h"
#include "dsp-protocol.h"
#include "constants.h"
#define ARRAY_SIZE(ary) (sizeof(ary)/sizeof(ary[0]))
/**
* Device node file name list.
*/
typedef struct {
char *device;
struct list_head list;
} device_list_t;
/**
* Holds the need information: list of playback and recording devices,
* current format, sample_rate, bytes per frame and pointer to ring
* buffer.
*/
typedef struct snd_pcm_alsa_dsp {
snd_pcm_ioplug_t io;
dsp_protocol_t *dsp_protocol;
int format;
int sample_rate;
int bytes_per_frame;
snd_pcm_sframes_t hw_pointer;
device_list_t playback_devices;
device_list_t recording_devices;
} snd_pcm_alsa_dsp_t;
static snd_pcm_alsa_dsp_t *free_ref;
/**
* @param io pcm io plugin configured to Alsa libs.
*
* It starts the playback sending a DSP_CMD_PLAY.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_start(snd_pcm_ioplug_t * io)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
int ret;
DENTER();
DPRINT("IO_STREAM %d == SND_PCM_STREAM_PLAYBACK %d\n", io->stream,
io->stream == SND_PCM_STREAM_PLAYBACK);
if (io->stream != SND_PCM_STREAM_PLAYBACK)
dsp_protocol_set_mic_enabled(alsa_dsp->dsp_protocol, 1);
ret = dsp_protocol_send_play(alsa_dsp->dsp_protocol);
DLEAVE(ret);
return ret;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* It starts the playback sending a DSP_CMD_STOP.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_stop(snd_pcm_ioplug_t * io)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
int ret;
DENTER();
ret = dsp_protocol_send_stop(alsa_dsp->dsp_protocol);
if (io->stream != SND_PCM_STREAM_PLAYBACK)
dsp_protocol_set_mic_enabled(alsa_dsp->dsp_protocol, 0);
DLEAVE(ret);
return ret;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* It returns the position of current period consuming.
*
* @return on success, returns current position, otherwise a negative
* error code.
*/
static snd_pcm_sframes_t alsa_dsp_pointer(snd_pcm_ioplug_t * io)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
snd_pcm_sframes_t ret;
DENTER();
ret = alsa_dsp->hw_pointer;
if (alsa_dsp->hw_pointer == 0)
alsa_dsp->hw_pointer =
io->period_size * alsa_dsp->bytes_per_frame;
else
alsa_dsp->hw_pointer = 0;
DLEAVE((int)ret);
return ret;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* It transfers the audio data to dsp side.
*
* @return on success, returns amount of data transfered,
* otherwise a negative error code.
*/
static snd_pcm_sframes_t alsa_dsp_transfer(snd_pcm_ioplug_t * io,
const snd_pcm_channel_area_t * areas,
snd_pcm_uframes_t offset,
snd_pcm_uframes_t size)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
DENTER();
char *buf;
int bytes, words;
ssize_t result;
bytes = size * alsa_dsp->bytes_per_frame;
DPRINT("***** Info: samples %lu * bpf %d => bytes %d\n",
size, alsa_dsp->bytes_per_frame, bytes);
if (bytes > alsa_dsp->dsp_protocol->mmap_buffer_size) {
DERROR("Requested too much data transfer (requested %d, playing only %d)\n",
bytes, alsa_dsp->dsp_protocol->mmap_buffer_size);
bytes = alsa_dsp->dsp_protocol->mmap_buffer_size;
}
words = bytes / 2;
if (alsa_dsp->dsp_protocol->state != STATE_PLAYING) {
DPRINT("I did nothing - No start sent\n");
alsa_dsp_start(io);
}
/* we handle only an interleaved buffer */
buf = (char *)areas->addr + (areas->first + areas->step * offset) / 8;
if (io->stream == SND_PCM_STREAM_PLAYBACK)
result =
dsp_protocol_send_audio_data(alsa_dsp->dsp_protocol, buf,
words);
else
result =
dsp_protocol_receive_audio_data(alsa_dsp->dsp_protocol, buf,
words);
result *= 2;
result /= alsa_dsp->bytes_per_frame;
alsa_dsp->hw_pointer += result;
DLEAVE(result);
return result;
}
/**
* @param device_list a list of device names to be freed.
*
* It passes a list of device names and frees each node.
*
* @return zero (success).
*/
static int free_device_list(device_list_t * device_list)
{
struct list_head *pos, *q;
device_list_t *tmp;
list_for_each_safe(pos, q, &device_list->list) {
tmp = list_entry(pos, device_list_t, list);
list_del(pos);
free(tmp->device);
free(tmp);
}
return 0;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* Closes the connection with the pcm dsp task. It
* destroies all allocated data.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_close(snd_pcm_ioplug_t * io)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
int ret = 0;
DENTER();
ret = dsp_protocol_close_node(alsa_dsp->dsp_protocol);
dsp_protocol_destroy(&(alsa_dsp->dsp_protocol));
free_device_list(&(alsa_dsp->playback_devices));
free_device_list(&(alsa_dsp->recording_devices));
DLEAVE(ret);
return ret;
}
/**
* @param map the values to be mapped
* @param value the search key
* @param steps how many keys should be checked
*
* Maps a value to another.
*
* @return on success, returns mapped value, otherwise a negative error code.
*/
static int map_value(int *map, int value, int steps)
{
int i;
for (i = 0; i < steps; i++)
if (map[i * 2] == value)
return map[i * 2 + 1];
return -1;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
* @param params
*
* It checks if the pcm format and rate are supported.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_hw_params(snd_pcm_ioplug_t * io,
snd_pcm_hw_params_t * params)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
int ret = 0;
int map_sample_rates[] = {
8000, SAMPLE_RATE_8KHZ,
11025, SAMPLE_RATE_11_025KHZ,
12000, SAMPLE_RATE_12KHZ,
16000, SAMPLE_RATE_16KHZ,
22050, SAMPLE_RATE_22_05KHZ,
24000, SAMPLE_RATE_24KHZ,
32000, SAMPLE_RATE_32KHZ,
44100, SAMPLE_RATE_44_1KHZ,
48000, SAMPLE_RATE_48KHZ
};
int map_formats[] = {
SND_PCM_FORMAT_A_LAW, DSP_AFMT_ALAW,
SND_PCM_FORMAT_MU_LAW, DSP_AFMT_ULAW,
SND_PCM_FORMAT_S16_LE, DSP_AFMT_S16_LE,
SND_PCM_FORMAT_U8, DSP_AFMT_U8,
SND_PCM_FORMAT_S8, DSP_AFMT_S8,
SND_PCM_FORMAT_S16_BE, DSP_AFMT_S16_BE,
SND_PCM_FORMAT_U16_LE, DSP_AFMT_U16_LE,
SND_PCM_FORMAT_U16_BE, DSP_AFMT_U16_BE
};
DENTER();
DPRINT("Checking Format- Ret %d\n", ret);
alsa_dsp->format = map_value(map_formats, io->format,
io->stream ==
SND_PCM_STREAM_PLAYBACK ?
ARRAY_SIZE(map_formats) : 3);
if (alsa_dsp->format < 0) {
DERROR("*** ALSA-DSP: unsupported format %s\n",
snd_pcm_format_name(io->format));
ret = -EINVAL;
}
DPRINT("Format is Ok. Checking rate. Ret %d\n", ret);
alsa_dsp->sample_rate = map_value(map_sample_rates, io->rate,
io->stream ==
SND_PCM_STREAM_PLAYBACK ?
ARRAY_SIZE(map_sample_rates) : 1);
if (alsa_dsp->sample_rate < 0) {
ret = -EINVAL;
DERROR("** ALSA - DSP - Unsuported Sample Rate! **\n");
}
DPRINT("Rate is ok. Calculating WPF. Ret %d\n", ret);
alsa_dsp->bytes_per_frame =
((snd_pcm_format_physical_width(io->format) * io->channels) / 8);
DPRINT("WPF: %d width %d channels %d\n", alsa_dsp->bytes_per_frame,
snd_pcm_format_physical_width(io->format), io->channels);
DLEAVE(ret);
return ret;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* It sends the audio parameters to pcm task node (formats, channels,
* access, rates). It is assumed that everything is proper set.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_prepare(snd_pcm_ioplug_t * io)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
audio_params_data_t params;
speech_params_data_t sparams;
int ret = 0;
char *tmp;
DENTER();
alsa_dsp->hw_pointer = 0;
if (alsa_dsp->dsp_protocol->state != STATE_INITIALISED) {
tmp = strdup(alsa_dsp->dsp_protocol->device);
ret = dsp_protocol_close_node(alsa_dsp->dsp_protocol);
if (!ret)
dsp_protocol_open_node(alsa_dsp->dsp_protocol, tmp);
free(tmp);
}
if (ret == 0) {
if (io->stream == SND_PCM_STREAM_PLAYBACK) {
params.dsp_cmd = DSP_CMD_SET_PARAMS;
params.dsp_audio_fmt = alsa_dsp->format;
params.sample_rate = alsa_dsp->sample_rate;
params.number_channels = io->channels;
params.ds_stream_id = 0;
params.stream_priority = 0;
if (dsp_protocol_send_audio_params
(alsa_dsp->dsp_protocol, ¶ms) < 0) {
ret = -EIO;
DERROR("Error in send params data\n");
} else
DPRINT("Sending params data is ok\n");
} else {
sparams.dsp_cmd = DSP_CMD_SET_SPEECH_PARAMS;
sparams.audio_fmt = alsa_dsp->format;
sparams.sample_rate = alsa_dsp->sample_rate;
sparams.ds_stream_id = 0;
sparams.stream_priority = 0;
sparams.frame_size = io->period_size;
DPRINT("frame size %u\n", sparams.frame_size);
if (dsp_protocol_send_speech_params
(alsa_dsp->dsp_protocol, &sparams) < 0) {
ret = -EIO;
DERROR("Error in send speech params data\n");
} else
DPRINT("Sending speech params data is ok\n");
}
}
DLEAVE(ret);
return ret;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* It pauses the playback sending a DSP_CMD_PAUSE.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_pause(snd_pcm_ioplug_t * io, int enable)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
int ret;
DENTER();
ret = dsp_protocol_send_pause(alsa_dsp->dsp_protocol);
DLEAVE(ret);
return ret;
}
/**
* @param io the pcm io plugin we configured to Alsa libs.
*
* It starts the playback sending a DSP_CMD_PLAY.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_resume(snd_pcm_ioplug_t * io)
{
snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data;
int ret;
DENTER();
ret = dsp_protocol_send_play(alsa_dsp->dsp_protocol);
DLEAVE(ret);
return ret;
}
/**
* @param alsa_dsp the structure to be configured.
*
* It configures constraints about formats, channels, access, rates,
* periods and buffer size. It exports the supported constraints by the
* dsp task node to the alsa plugin library.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_configure_constraints(snd_pcm_alsa_dsp_t * alsa_dsp)
{
snd_pcm_ioplug_t *io = &alsa_dsp->io;
static const snd_pcm_access_t access_list[] = {
SND_PCM_ACCESS_RW_INTERLEAVED
};
static const unsigned int formats[] = {
SND_PCM_FORMAT_U8, /* DSP_AFMT_U8 */
SND_PCM_FORMAT_S16_LE, /* DSP_AFMT_S16_LE */
SND_PCM_FORMAT_S16_BE, /* DSP_AFMT_S16_BE */
SND_PCM_FORMAT_S8, /* DSP_AFMT_S8 */
SND_PCM_FORMAT_U16_LE, /* DSP_AFMT_U16_LE */
SND_PCM_FORMAT_U16_BE, /* DSP_AFMT_U16_BE */
SND_PCM_FORMAT_A_LAW, /* DSP_AFMT_ALAW */
SND_PCM_FORMAT_MU_LAW /* DSP_AFMT_ULAW */
};
static const unsigned int formats_recor[] = {
SND_PCM_FORMAT_S16_LE, /* DSP_AFMT_S16_LE */
SND_PCM_FORMAT_A_LAW, /* DSP_AFMT_ALAW */
SND_PCM_FORMAT_MU_LAW /* DSP_AFMT_ULAW */
};
static const unsigned int bytes_list[] = {
1U << 11, 1U << 12
};
static const unsigned int bytes_list_rec_8bit[] = {
/* It must be multiple of 80... less than or equal to 800 */
80, 160, 240, 320, 400, 480, 560, 640, 720, 800
};
int ret, err;
DENTER();
/* Configuring access */
if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS,
ARRAY_SIZE(access_list),
access_list)) < 0) {
ret = err;
goto out;
}
if (io->stream == SND_PCM_STREAM_PLAYBACK) {
/* Configuring formats */
if ((err =
snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT,
ARRAY_SIZE(formats),
formats)) < 0) {
ret = err;
goto out;
}
/* Configuring channels */
if ((err =
snd_pcm_ioplug_set_param_minmax(io,
SND_PCM_IOPLUG_HW_CHANNELS,
1, 2)) < 0) {
ret = err;
goto out;
}
/* Configuring rates */
if ((err =
snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE,
8000, 48000)) < 0) {
ret = err;
goto out;
}
/* Configuring periods */
if ((err =
snd_pcm_ioplug_set_param_list(io,
SND_PCM_IOPLUG_HW_PERIOD_BYTES,
ARRAY_SIZE(bytes_list),
bytes_list)) < 0) {
ret = err;
goto out;
}
/* Configuring buffer size */
if ((err =
snd_pcm_ioplug_set_param_list(io,
SND_PCM_IOPLUG_HW_BUFFER_BYTES,
ARRAY_SIZE(bytes_list),
bytes_list)) < 0) {
ret = err;
goto out;
}
} else {
/* Configuring formats */
if ((err =
snd_pcm_ioplug_set_param_list(io,
SND_PCM_IOPLUG_HW_FORMAT,
ARRAY_SIZE(formats_recor),
formats_recor)) < 0) {
ret = err;
goto out;
}
/* Configuring channels */
if ((err = snd_pcm_ioplug_set_param_minmax(io,
SND_PCM_IOPLUG_HW_CHANNELS,
1, 1)) < 0) {
ret = err;
goto out;
}
/* Configuring rates */
if ((err =
snd_pcm_ioplug_set_param_minmax(io,
SND_PCM_IOPLUG_HW_RATE,
8000, 8000)) < 0) {
ret = err;
goto out;
}
/* Configuring periods */
if ((err =
snd_pcm_ioplug_set_param_list(io,
SND_PCM_IOPLUG_HW_PERIOD_BYTES,
ARRAY_SIZE
(bytes_list_rec_8bit),
bytes_list_rec_8bit)) < 0) {
ret = err;
goto out;
}
/* Configuring buffer size */
if ((err =
snd_pcm_ioplug_set_param_list(io,
SND_PCM_IOPLUG_HW_BUFFER_BYTES,
ARRAY_SIZE
(bytes_list_rec_8bit),
bytes_list_rec_8bit)) < 0) {
ret = err;
goto out;
}
}
if ((err = snd_pcm_ioplug_set_param_minmax(io,
SND_PCM_IOPLUG_HW_PERIODS,
2, 1024)) < 0) {
ret = err;
goto out;
}
ret = 0;
out:
DLEAVE(ret);
return ret;
}
/**
* Alsa-lib callback structure.
*/
static snd_pcm_ioplug_callback_t alsa_dsp_callback = {
.start = alsa_dsp_start,
.stop = alsa_dsp_stop,
.pointer = alsa_dsp_pointer,
.transfer = alsa_dsp_transfer,
.close = alsa_dsp_close,
.hw_params = alsa_dsp_hw_params,
.prepare = alsa_dsp_prepare,
.pause = alsa_dsp_pause,
.resume = alsa_dsp_resume,
};
/**
* @param alsa_dsp the structure to be configured.
*
* It probes all configured dsp task devices to be available for
* this plugin. It will use first dsp task device whose is in
* UNINITIALISED state.
*
* @return zero if success, otherwise a negative error code.
*/
static int alsa_dsp_open_dsp_task(snd_pcm_alsa_dsp_t * alsa_dsp,
device_list_t * device_list)
{
int err = -EINVAL;
device_list_t *tmp;
DENTER();
DPRINT("Looking for a dsp device node \n");
list_for_each_entry(tmp, &(device_list->list), list) {
DPRINT("Trying to use %s\n", tmp->device);
if ((err =
dsp_protocol_open_node(alsa_dsp->dsp_protocol,
tmp->device)) < 0) {
DPRINT("%s is not available now\n", tmp->device);
dsp_protocol_close_node(alsa_dsp->dsp_protocol);
} else
break;
}
if (err < 0) {
DPRINT("No valid dsp task nodes for now. Exiting.\n");
}
DLEAVE(err);
return err;
}
/**
* @param n configuration file parse tree.
* @param device_list list of device files to be filled.
*
* It searches for device file names in given configuration parse
* tree. When one device file name is found, it is filled into device_list.
*
* @return zero if success, otherwise a negative error code.
*/
static int fill_string_list(snd_config_t * n, device_list_t * device_list)
{
snd_config_iterator_t j, nextj;
device_list_t *tmp;
long idx = 0;
int ret;
DENTER();
INIT_LIST_HEAD(&device_list->list);
snd_config_for_each(j, nextj, n) {
snd_config_t *s = snd_config_iterator_entry(j);
const char *id_number;
long k;
if (snd_config_get_id(s, &id_number) < 0)
continue;
if (safe_strtol(id_number, &k) < 0) {
SNDERR("id of field %s is not an integer", id_number);
ret = -EINVAL;
goto out;
}
if (k == idx) {
idx++;
/* add to available dsp task nodes */
tmp = (device_list_t *) malloc(sizeof(device_list_t));
if (snd_config_get_ascii(s, &(tmp->device)) < 0) {
SNDERR("invalid ascii string for id %s\n",
id_number);
ret = -EINVAL;
goto out;
}
list_add(&(tmp->list), &(device_list->list));
}
}
ret = 0;
out:
DLEAVE(ret);
return ret;
}
/**
* It initializes the alsa plugin. It reads the parameters and creates the
* connection with the pcm device file.
*
* @return zero if success, otherwise a negative error code.
*/
SND_PCM_PLUGIN_DEFINE_FUNC(alsa_dsp)
{
snd_config_iterator_t i, next;
snd_pcm_alsa_dsp_t *alsa_dsp;
int err;
int ret;
DENTER();
/* Allocate the structure */
alsa_dsp = calloc(1, sizeof(snd_pcm_alsa_dsp_t));
if (alsa_dsp == NULL) {
ret = -ENOMEM;
goto out;
}
/* Read the configuration searching for configurated devices */
snd_config_for_each(i, next, conf) {
snd_config_t *n = snd_config_iterator_entry(i);
const char *id;
if (snd_config_get_id(n, &id) < 0)
continue;
if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0)
continue;
if (strcmp(id, "playback_device_file") == 0) {
if (snd_config_get_type(n) == SND_CONFIG_TYPE_COMPOUND){
if ((err =
fill_string_list(n,
&(alsa_dsp->playback_devices))) < 0) {
SNDERR("Could not fill string"
" list for playback devices\n");
goto error;
}
} else {
SNDERR("Invalid type for %s", id);
err = -EINVAL;
goto error;
}
continue;
}
if (strcmp(id, "recording_device_file") == 0) {
if (snd_config_get_type(n) == SND_CONFIG_TYPE_COMPOUND){
if ((err =
fill_string_list(n,
&(alsa_dsp->recording_devices))) < 0){
SNDERR("Could not fill string"
" list for recording devices\n");
goto error;
}
} else {
SNDERR("Invalid type for %s", id);
err = -EINVAL;
goto error;
}
continue;
}
SNDERR("Unknown field %s", id);
err = -EINVAL;
goto error;
}
/* Initialise the dsp_protocol and create connection */
if ((err = dsp_protocol_create(&(alsa_dsp->dsp_protocol))) < 0)
goto error;
if ((err = alsa_dsp_open_dsp_task(alsa_dsp,
(stream == SND_PCM_STREAM_PLAYBACK) ?
&(alsa_dsp->playback_devices) :
&(alsa_dsp->recording_devices))) < 0)
goto error;
/* Initialise the snd_pcm_ioplug_t */
alsa_dsp->io.version = SND_PCM_IOPLUG_VERSION;
alsa_dsp->io.name = "Alsa - DSP PCM Plugin";
alsa_dsp->io.mmap_rw = 0;
alsa_dsp->io.callback = &alsa_dsp_callback;
alsa_dsp->io.poll_fd = alsa_dsp->dsp_protocol->fd;
alsa_dsp->io.poll_events = stream == SND_PCM_STREAM_PLAYBACK ?
POLLOUT : POLLIN;
alsa_dsp->io.private_data = alsa_dsp;
free_ref = alsa_dsp;
if ((err = snd_pcm_ioplug_create(&alsa_dsp->io, name,
stream, mode)) < 0)
goto error;
/* Configure the plugin */
if ((err = alsa_dsp_configure_constraints(alsa_dsp)) < 0) {
snd_pcm_ioplug_delete(&alsa_dsp->io);
goto error;
}
*pcmp = alsa_dsp->io.pcm;
ret = 0;
goto out;
error:
ret = err;
free(alsa_dsp);
out:
DLEAVE(ret);
return ret;
}
static void alsa_dsp_descructor(void) __attribute__ ((destructor));
static void alsa_dsp_descructor(void)
{
DENTER();
DPRINT("alsa dsp destructor\n");
DPRINT("checking for memories leaks and releasing resources\n");
if (free_ref) {
if (free_ref->dsp_protocol) {
dsp_protocol_close_node(free_ref->dsp_protocol);
dsp_protocol_destroy(&(free_ref->dsp_protocol));
}
free_device_list(&(free_ref->playback_devices));
free_device_list(&(free_ref->recording_devices));
free(free_ref);
free_ref = NULL;
}
DLEAVE(0);
}
SND_PCM_PLUGIN_SYMBOL(alsa_dsp);