/* GStreamer

 * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>

 * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>

 *

 * gstalsasink.c:

 *

 * 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:element-alsasink

 * @title: alsasink

 * @see_also: alsasrc

 *

 * This element renders audio samples using the ALSA audio API.

 *

 * ## Example pipelines

 * |[

 * gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! autoaudiosink

 * ]|

 *

 * Play an Ogg/Vorbis file and output audio via ALSA.

 *

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <sys/ioctl.h>

#include <fcntl.h>

#include <errno.h>

#include <unistd.h>

#include <string.h>

#include <getopt.h>

#include <alsa/asoundlib.h>

#include "gstalsa.h"

#include "gstalsasink.h"

#include "gstalsadeviceprobe.h"

#include <gst/audio/gstaudioiec61937.h>

#include <gst/gst-i18n-plugin.h>

#ifndef ESTRPIPE

#define ESTRPIPE EPIPE

#endif

#define DEFAULT_DEVICE		"default"

#define DEFAULT_DEVICE_NAME	""

#define DEFAULT_CARD_NAME	""

#define SPDIF_PERIOD_SIZE 1536

#define SPDIF_BUFFER_SIZE 15360

enum

{

  PROP_0,

  PROP_DEVICE,

  PROP_DEVICE_NAME,

  PROP_CARD_NAME,

  PROP_LAST

};

static void gst_alsasink_init_interfaces (GType type);

#define gst_alsasink_parent_class parent_class

G_DEFINE_TYPE_WITH_CODE (GstAlsaSink, gst_alsasink,

    GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces (g_define_type_id));

static void gst_alsasink_finalise (GObject * object);

static void gst_alsasink_set_property (GObject * object,

    guint prop_id, const GValue * value, GParamSpec * pspec);

static void gst_alsasink_get_property (GObject * object,

    guint prop_id, GValue * value, GParamSpec * pspec);

static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter);

static gboolean gst_alsasink_query (GstBaseSink * bsink, GstQuery * query);

static gboolean gst_alsasink_open (GstAudioSink * asink);

static gboolean gst_alsasink_prepare (GstAudioSink * asink,

    GstAudioRingBufferSpec * spec);

static gboolean gst_alsasink_unprepare (GstAudioSink * asink);

static gboolean gst_alsasink_close (GstAudioSink * asink);

static gint gst_alsasink_write (GstAudioSink * asink, gpointer data,

    guint length);

static guint gst_alsasink_delay (GstAudioSink * asink);

static void gst_alsasink_reset (GstAudioSink * asink);

static gboolean gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps);

static GstBuffer *gst_alsasink_payload (GstAudioBaseSink * sink,

    GstBuffer * buf);

static gint output_ref;         /* 0    */

static snd_output_t *output;    /* NULL */

static GMutex output_mutex;

static GstStaticPadTemplate alsasink_sink_factory =

    GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-raw, "

        "format = (string) " GST_AUDIO_FORMATS_ALL ", "

        "layout = (string) interleaved, "

        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "

        PASSTHROUGH_CAPS)

    );

static void

gst_alsasink_finalise (GObject * object)

{

  GstAlsaSink *sink = GST_ALSA_SINK (object);

  g_free (sink->device);

  g_mutex_clear (&sink->alsa_lock);

  g_mutex_clear (&sink->delay_lock);

  g_mutex_lock (&output_mutex);

  --output_ref;

  if (output_ref == 0) {

    snd_output_close (output);

    output = NULL;

  }

  g_mutex_unlock (&output_mutex);

  G_OBJECT_CLASS (parent_class)->finalize (object);

}

static void

gst_alsasink_init_interfaces (GType type)

{

#if 0

  gst_alsa_type_add_device_property_probe_interface (type);

#endif

}

static void

gst_alsasink_class_init (GstAlsaSinkClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstBaseSinkClass *gstbasesink_class;

  GstAudioBaseSinkClass *gstbaseaudiosink_class;

  GstAudioSinkClass *gstaudiosink_class;

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstbasesink_class = (GstBaseSinkClass *) klass;

  gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;

  gstaudiosink_class = (GstAudioSinkClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  gobject_class->finalize = gst_alsasink_finalise;

  gobject_class->get_property = gst_alsasink_get_property;

  gobject_class->set_property = gst_alsasink_set_property;

  gst_element_class_set_static_metadata (gstelement_class,

      "Audio sink (ALSA)", "Sink/Audio",

      "Output to a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");

  gst_element_class_add_static_pad_template (gstelement_class,

      &alsasink_sink_factory);

  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);

  gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_alsasink_query);

  gstbaseaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_alsasink_payload);

  gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);

  gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);

  gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);

  gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);

  gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);

  gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);

  gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);

  g_object_class_install_property (gobject_class, PROP_DEVICE,

      g_param_spec_string ("device", "Device",

          "ALSA device, as defined in an asound configuration file",

          DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,

      g_param_spec_string ("device-name", "Device name",

          "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,

          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_CARD_NAME,

      g_param_spec_string ("card-name", "Card name",

          "Human-readable name of the sound card", DEFAULT_CARD_NAME,

          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

}

static void

gst_alsasink_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec)

{

  GstAlsaSink *sink;

  sink = GST_ALSA_SINK (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_free (sink->device);

      sink->device = g_value_dup_string (value);

      /* setting NULL restores the default device */

      if (sink->device == NULL) {

        sink->device = g_strdup (DEFAULT_DEVICE);

      }

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_alsasink_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec)

{

  GstAlsaSink *sink;

  sink = GST_ALSA_SINK (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_value_set_string (value, sink->device);

      break;

    case PROP_DEVICE_NAME:

      g_value_take_string (value,

          gst_alsa_find_device_name (GST_OBJECT_CAST (sink),

              sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));

      break;

    case PROP_CARD_NAME:

      g_value_take_string (value,

          gst_alsa_find_card_name (GST_OBJECT_CAST (sink),

              sink->device, SND_PCM_STREAM_PLAYBACK));

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_alsasink_init (GstAlsaSink * alsasink)

{

  GST_DEBUG_OBJECT (alsasink, "initializing alsasink");

  alsasink->device = g_strdup (DEFAULT_DEVICE);

  alsasink->handle = NULL;

  alsasink->cached_caps = NULL;

  g_mutex_init (&alsasink->alsa_lock);

  g_mutex_init (&alsasink->delay_lock);

  g_mutex_lock (&output_mutex);

  if (output_ref == 0) {

    snd_output_stdio_attach (&output, stdout, 0);

    ++output_ref;

  }

  g_mutex_unlock (&output_mutex);

}

#define CHECK(call, error) \

G_STMT_START {             \

  if ((err = call) < 0) {  \

    GST_WARNING_OBJECT (alsa, "Error %d (%s) calling " #call, err, snd_strerror (err)); \

    goto error;            \

  }                        \

} G_STMT_END;

static GstCaps *

gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter)

{

  GstElementClass *element_class;

  GstPadTemplate *pad_template;

  GstAlsaSink *sink = GST_ALSA_SINK (bsink);

  GstCaps *caps, *templ_caps;

  GST_OBJECT_LOCK (sink);

  if (sink->handle == NULL) {

    GST_OBJECT_UNLOCK (sink);

    GST_DEBUG_OBJECT (sink, "device not open, using template caps");

    return NULL;                /* base class will get template caps for us */

  }

  if (sink->cached_caps) {

    if (filter) {

      caps = gst_caps_intersect_full (filter, sink->cached_caps,

          GST_CAPS_INTERSECT_FIRST);

      GST_OBJECT_UNLOCK (sink);

      GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT " with "

          "filter %" GST_PTR_FORMAT " applied: %" GST_PTR_FORMAT,

          sink->cached_caps, filter, caps);

      return caps;

    } else {

      caps = gst_caps_ref (sink->cached_caps);

      GST_OBJECT_UNLOCK (sink);

      GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT, caps);

      return caps;

    }

  }

  element_class = GST_ELEMENT_GET_CLASS (sink);

  pad_template = gst_element_class_get_pad_template (element_class, "sink");

  if (pad_template == NULL) {

    GST_OBJECT_UNLOCK (sink);

    g_assert_not_reached ();

    return NULL;

  }

  templ_caps = gst_pad_template_get_caps (pad_template);

  caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->device,

      sink->handle, templ_caps);

  gst_caps_unref (templ_caps);

  if (caps) {

    sink->cached_caps = gst_caps_ref (caps);

  }

  GST_OBJECT_UNLOCK (sink);

  GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);

  if (filter) {

    GstCaps *intersection;

    intersection =

        gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (caps);

    return intersection;

  } else {

    return caps;

  }

}

static gboolean

gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps)

{

  GstPad *pad = GST_BASE_SINK (alsa)->sinkpad;

  GstCaps *pad_caps;

  GstStructure *st;

  gboolean ret = FALSE;

  GstAudioRingBufferSpec spec = { 0 };

  pad_caps = gst_pad_query_caps (pad, caps);

  if (!pad_caps || gst_caps_is_empty (pad_caps)) {

    if (pad_caps)

      gst_caps_unref (pad_caps);

    ret = FALSE;

    goto done;

  }

  gst_caps_unref (pad_caps);

  /* If we've not got fixed caps, creating a stream might fail, so let's just

   * return from here with default acceptcaps behaviour */

  if (!gst_caps_is_fixed (caps))

    goto done;

  /* parse helper expects this set, so avoid nasty warning

   * will be set properly later on anyway  */

  spec.latency_time = GST_SECOND;

  if (!gst_audio_ring_buffer_parse_caps (&spec, caps))

    goto done;

  /* Make sure input is framed (one frame per buffer) and can be payloaded */

  switch (spec.type) {

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:

    {

      gboolean framed = FALSE, parsed = FALSE;

      st = gst_caps_get_structure (caps, 0);

      gst_structure_get_boolean (st, "framed", &framed);

      gst_structure_get_boolean (st, "parsed", &parsed);

      if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)

        goto done;

    }

    default:{

    }

  }

  ret = TRUE;

done:

  gst_caps_replace (&spec.caps, NULL);

  return ret;

}

static gboolean

gst_alsasink_query (GstBaseSink * sink, GstQuery * query)

{

  GstAlsaSink *alsa = GST_ALSA_SINK (sink);

  gboolean ret;

  switch (GST_QUERY_TYPE (query)) {

    case GST_QUERY_ACCEPT_CAPS:

    {

      GstCaps *caps;

      gst_query_parse_accept_caps (query, &caps);

      ret = gst_alsasink_acceptcaps (alsa, caps);

      gst_query_set_accept_caps_result (query, ret);

      ret = TRUE;

      break;

    }

    default:

      ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);

      break;

  }

  return ret;

}

static int

set_hwparams (GstAlsaSink * alsa)

{

  guint rrate;

  gint err;

  snd_pcm_hw_params_t *params;

  guint period_time, buffer_time;

  snd_pcm_hw_params_malloc (&params);

  GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "

      "SPDIF (%d)", alsa->channels, alsa->rate,

      snd_pcm_format_name (alsa->format), alsa->iec958);

  /* start with requested values, if we cannot configure alsa for those values,

   * we set these values to -1, which will leave the default alsa values */

  buffer_time = alsa->buffer_time;

  period_time = alsa->period_time;

retry:

  /* choose all parameters */

  CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);

  /* set the interleaved read/write format */

  CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),

      wrong_access);

  /* set the sample format */

  if (alsa->iec958) {

    /* Try to use big endian first else fallback to le and swap bytes */

    if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {

      alsa->format = SND_PCM_FORMAT_S16_LE;

      alsa->need_swap = TRUE;

      GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");

    } else {

      alsa->need_swap = FALSE;

    }

  }

  CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),

      no_sample_format);

  /* set the count of channels */

  CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),

      no_channels);

  /* set the stream rate */

  rrate = alsa->rate;

  CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),

      no_rate);

#ifndef GST_DISABLE_GST_DEBUG

  /* get and dump some limits */

  {

    guint min, max;

    snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);

    snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);

    GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",

        alsa->buffer_time, min, max);

    snd_pcm_hw_params_get_period_time_min (params, &min, NULL);

    snd_pcm_hw_params_get_period_time_max (params, &max, NULL);

    GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",

        alsa->period_time, min, max);

    snd_pcm_hw_params_get_periods_min (params, &min, NULL);

    snd_pcm_hw_params_get_periods_max (params, &max, NULL);

    GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);

  }

#endif

  /* now try to configure the buffer time and period time, if one

   * of those fail, we fall back to the defaults and emit a warning. */

  if (buffer_time != -1 && !alsa->iec958) {

    /* set the buffer time */

    if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,

                &buffer_time, NULL)) < 0) {

      GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),

          ("Unable to set buffer time %i for playback: %s",

              buffer_time, snd_strerror (err)));

      /* disable buffer_time the next round */

      buffer_time = -1;

      goto retry;

    }

    GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);

    alsa->buffer_time = buffer_time;

  }

  if (period_time != -1 && !alsa->iec958) {

    /* set the period time */

    if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,

                &period_time, NULL)) < 0) {

      GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),

          ("Unable to set period time %i for playback: %s",

              period_time, snd_strerror (err)));

      /* disable period_time the next round */

      period_time = -1;

      goto retry;

    }

    GST_DEBUG_OBJECT (alsa, "period time %u", period_time);

    alsa->period_time = period_time;

  }

  /* Set buffer size and period size manually for SPDIF */

  if (G_UNLIKELY (alsa->iec958)) {

    snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;

    snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;

    CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,

            &buffer_size), buffer_size);

    CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,

            &period_size, NULL), period_size);

  }

  /* write the parameters to device */

  CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);

  /* now get the configured values */

  CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),

      buffer_size);

  CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),

      period_size);

  GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,

      alsa->period_size);

  snd_pcm_hw_params_free (params);

  return 0;

  /* ERRORS */

no_config:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Broken configuration for playback: no configurations available: %s",

            snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

wrong_access:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Access type not available for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

no_sample_format:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Sample format not available for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

no_channels:

  {

    gchar *msg = NULL;

    if ((alsa->channels) == 1)

      msg = g_strdup (_("Could not open device for playback in mono mode."));

    if ((alsa->channels) == 2)

      msg = g_strdup (_("Could not open device for playback in stereo mode."));

    if ((alsa->channels) > 2)

      msg =

          g_strdup_printf (_

          ("Could not open device for playback in %d-channel mode."),

          alsa->channels);

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),

        ("%s", snd_strerror (err)));

    g_free (msg);

    snd_pcm_hw_params_free (params);

    return err;

  }

no_rate:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Rate %iHz not available for playback: %s",

            alsa->rate, snd_strerror (err)));

    return err;

  }

buffer_size:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to get buffer size for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

period_size:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to get period size for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

set_hw_params:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to set hw params for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

}

static int

set_swparams (GstAlsaSink * alsa)

{

  int err;

  snd_pcm_sw_params_t *params;

  snd_pcm_sw_params_malloc (&params);

  /* get the current swparams */

  CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);

  /* start the transfer when the buffer is almost full: */

  /* (buffer_size / avail_min) * avail_min */

  CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,

          (alsa->buffer_size / alsa->period_size) * alsa->period_size),

      start_threshold);

  /* allow the transfer when at least period_size samples can be processed */

  CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,

          alsa->period_size), set_avail);

#if GST_CHECK_ALSA_VERSION(1,0,16)

  /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */

#else

  /* align all transfers to 1 sample */

  CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);

#endif

  /* write the parameters to the playback device */

  CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);

  snd_pcm_sw_params_free (params);

  return 0;

  /* ERRORS */

no_config:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to determine current swparams for playback: %s",

            snd_strerror (err)));

    snd_pcm_sw_params_free (params);

    return err;

  }

start_threshold:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to set start threshold mode for playback: %s",

            snd_strerror (err)));

    snd_pcm_sw_params_free (params);

    return err;

  }

set_avail:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to set avail min for playback: %s", snd_strerror (err)));

    snd_pcm_sw_params_free (params);

    return err;

  }

#if !GST_CHECK_ALSA_VERSION(1,0,16)

set_align:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to set transfer align for playback: %s", snd_strerror (err)));

    snd_pcm_sw_params_free (params);

    return err;

  }

#endif

set_sw_params:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to set sw params for playback: %s", snd_strerror (err)));

    snd_pcm_sw_params_free (params);

    return err;

  }

}

static gboolean

alsasink_parse_spec (GstAlsaSink * alsa, GstAudioRingBufferSpec * spec)

{

  /* Initialize our boolean */

  alsa->iec958 = FALSE;

  switch (spec->type) {

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:

      switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {

        case GST_AUDIO_FORMAT_U8:

          alsa->format = SND_PCM_FORMAT_U8;

          break;

        case GST_AUDIO_FORMAT_S8:

          alsa->format = SND_PCM_FORMAT_S8;

          break;

        case GST_AUDIO_FORMAT_S16LE:

          alsa->format = SND_PCM_FORMAT_S16_LE;

          break;

        case GST_AUDIO_FORMAT_S16BE:

          alsa->format = SND_PCM_FORMAT_S16_BE;

          break;

        case GST_AUDIO_FORMAT_U16LE:

          alsa->format = SND_PCM_FORMAT_U16_LE;

          break;

        case GST_AUDIO_FORMAT_U16BE:

          alsa->format = SND_PCM_FORMAT_U16_BE;

          break;

        case GST_AUDIO_FORMAT_S24_32LE:

          alsa->format = SND_PCM_FORMAT_S24_LE;

          break;

        case GST_AUDIO_FORMAT_S24_32BE:

          alsa->format = SND_PCM_FORMAT_S24_BE;

          break;

        case GST_AUDIO_FORMAT_U24_32LE:

          alsa->format = SND_PCM_FORMAT_U24_LE;

          break;

        case GST_AUDIO_FORMAT_U24_32BE:

          alsa->format = SND_PCM_FORMAT_U24_BE;

          break;

        case GST_AUDIO_FORMAT_S32LE:

          alsa->format = SND_PCM_FORMAT_S32_LE;

          break;

        case GST_AUDIO_FORMAT_S32BE:

          alsa->format = SND_PCM_FORMAT_S32_BE;

          break;

        case GST_AUDIO_FORMAT_U32LE:

          alsa->format = SND_PCM_FORMAT_U32_LE;

          break;

        case GST_AUDIO_FORMAT_U32BE:

          alsa->format = SND_PCM_FORMAT_U32_BE;

          break;

        case GST_AUDIO_FORMAT_S24LE:

          alsa->format = SND_PCM_FORMAT_S24_3LE;

          break;

        case GST_AUDIO_FORMAT_S24BE:

          alsa->format = SND_PCM_FORMAT_S24_3BE;

          break;

        case GST_AUDIO_FORMAT_U24LE:

          alsa->format = SND_PCM_FORMAT_U24_3LE;

          break;

        case GST_AUDIO_FORMAT_U24BE:

          alsa->format = SND_PCM_FORMAT_U24_3BE;

          break;

        case GST_AUDIO_FORMAT_S20LE:

          alsa->format = SND_PCM_FORMAT_S20_3LE;

          break;

        case GST_AUDIO_FORMAT_S20BE:

          alsa->format = SND_PCM_FORMAT_S20_3BE;

          break;

        case GST_AUDIO_FORMAT_U20LE:

          alsa->format = SND_PCM_FORMAT_U20_3LE;

          break;

        case GST_AUDIO_FORMAT_U20BE:

          alsa->format = SND_PCM_FORMAT_U20_3BE;

          break;

        case GST_AUDIO_FORMAT_S18LE:

          alsa->format = SND_PCM_FORMAT_S18_3LE;

          break;

        case GST_AUDIO_FORMAT_S18BE:

          alsa->format = SND_PCM_FORMAT_S18_3BE;

          break;

        case GST_AUDIO_FORMAT_U18LE:

          alsa->format = SND_PCM_FORMAT_U18_3LE;

          break;

        case GST_AUDIO_FORMAT_U18BE:

          alsa->format = SND_PCM_FORMAT_U18_3BE;

          break;

        case GST_AUDIO_FORMAT_F32LE:

          alsa->format = SND_PCM_FORMAT_FLOAT_LE;

          break;

        case GST_AUDIO_FORMAT_F32BE:

          alsa->format = SND_PCM_FORMAT_FLOAT_BE;

          break;

        case GST_AUDIO_FORMAT_F64LE:

          alsa->format = SND_PCM_FORMAT_FLOAT64_LE;

          break;

        case GST_AUDIO_FORMAT_F64BE:

          alsa->format = SND_PCM_FORMAT_FLOAT64_BE;

          break;

        default:

          goto error;

      }

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:

      alsa->format = SND_PCM_FORMAT_A_LAW;

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:

      alsa->format = SND_PCM_FORMAT_MU_LAW;

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:

      alsa->format = SND_PCM_FORMAT_S16_BE;

      alsa->iec958 = TRUE;

      break;

    default:

      goto error;

  }

  alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);

  alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);

  alsa->buffer_time = spec->buffer_time;

  alsa->period_time = spec->latency_time;

  alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;

  if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)

    gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK

        (alsa)->ringbuffer, alsa_position[alsa->channels - 1]);

  return TRUE;

  /* ERRORS */

error:

  {

    return FALSE;

  }

}

static gboolean

gst_alsasink_open (GstAudioSink * asink)

{

  GstAlsaSink *alsa;

  gint err;

  alsa = GST_ALSA_SINK (asink);

  /* open in non-blocking mode, we'll use snd_pcm_wait() for space to become

   * available. */

  CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,

          SND_PCM_NONBLOCK), open_error);

  GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);

  return TRUE;

  /* ERRORS */

open_error:

  {

    if (err == -EBUSY) {

      GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,

          (_("Could not open audio device for playback. "

                  "Device is being used by another application.")),

          ("Device '%s' is busy", alsa->device));

    } else {

      GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,

          (_("Could not open audio device for playback.")),

          ("Playback open error on device '%s': %s", alsa->device,

              snd_strerror (err)));

    }

    return FALSE;

  }

}

static gboolean

gst_alsasink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)

{

  GstAlsaSink *alsa;

  gint err;

  alsa = GST_ALSA_SINK (asink);

  if (alsa->iec958) {

    snd_pcm_close (alsa->handle);