/**

 * @file alsa-dsp.c

 * @brief Alsa External plugin: I/O plugin

 * 

 * Copyright (C) 2006 Nokia Corporation

 * 

 * 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, &params) < 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);