This patch file adds an ALSA External PCM I/O plugin. This source uses
the dsp-protocol
implementation.
The plugin probes for a free communication channel at the start time.
It will probe only
for channels specified into the configuration file for the plugin. An
configuration example is:
# PCM
pcm.!default {
type alsa_dsp
playback_device_file ["/dev/dsptask/pcm2"]
recording_device_file ["/dev/dsptask/pcm_rec"]
}
The plugin supports the following:
* Playback:
o 16-bit PCM formats:
+ S16_LE
+ S16_BE
+ U16_LE
+ U16_BE
o 8-bit PCM formats:
+ A_LAW
+ MU_LAW
+ U8
+ S8
o Rates:
+ 8 KHz
+ 11.025 KHz
+ 12 KHz
+ 16 KHz
+ 22.050 KHz
+ 24 KHz
+ 32 KHz
+ 44.1 KHz
+ 48 KHz
o Channels:
+ Mono
+ Stereo
* Recording:
o 16-bit PCM formats:
+ S16_LE
o 8-bit PCM formats:
+ A_LAW
+ MU_LAW
o Rates:
+ 8 KHz
o Channels
+ Mono
Signed-off-by: Eduardo Valentin <eduardo.valentin@xxxxxxxxxxx>
diffstat:
alsa-dsp.c | 772 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
1 files changed, 772 insertions(+)
--
Eduardo Bezerra Valentin
diff -pruN a/maemo/alsa-dsp.c b/maemo/alsa-dsp.c
--- a/maemo/alsa-dsp.c 1969-12-31 20:00:00.000000000 -0400
+++ b/maemo/alsa-dsp.c 2006-11-01 11:55:58.000000000 -0400
@@ -0,0 +1,772 @@
+/**
+ * @file alsa-dsp.c
+ * @brief Alsa External plugin: I/O plugin
+ * <p>
+ * Copyright (C) 2006 Nokia Corporation
+ * <p>
+ * Contact: Eduardo Bezerra Valentin <eduardo.valentin@xxxxxxxxxxx>
+ *
+ * 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 library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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 words;
+ ssize_t result;
+
+ words = size * alsa_dsp->bytes_per_frame;
+ words /= 2;
+ DPRINT("***** Info: words %d size %lu bpf: %d\n", words, size,
+ alsa_dsp->bytes_per_frame);
+ if (words > alsa_dsp->dsp_protocol->mmap_buffer_size) {
+ DERROR("Requested too much data transfer (playing only %d)\n",
+ alsa_dsp->dsp_protocol->mmap_buffer_size);
+ words = alsa_dsp->dsp_protocol->mmap_buffer_size;
+ }
+ 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;
+ out:
+ 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 snd_pcm_access_t access_list[] = {
+ SND_PCM_ACCESS_RW_INTERLEAVED
+ };
+ 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 */
+ };
+ 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 unsigned int bytes_list[] = {
+ 1U << 11, 1U << 12
+ };
+ static 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)
+ 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;
+}
+
+
+void alsa_dsp_descructor(void) __attribute__ ((destructor));
+
+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);
-------------------------------------------------------------------------
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