/* GStreamer
 * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
 *
 * 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., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

/**
 * SECTION:element-alsasink
 * @short_description: play audio to an ALSA device
 * @see_also: alsasrc, alsamixer
 *
 * <refsect2>
 * <para>
 * This element renders raw audio samples using the ALSA api.
 * </para>
 * <title>Example pipelines</title>
 * <para>
 * Play an Ogg/Vorbis file.
 * </para>
 * <programlisting>
 * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
 * </programlisting>
 * </refsect2>
 *
 * Last reviewed on 2006-03-01 (0.10.4)
 */

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

/* elementfactory information */
static const GstElementDetails gst_alsasink_details =
GST_ELEMENT_DETAILS ("Audio sink (ALSA)",
    "Sink/Audio",
    "Output to a sound card via ALSA",
    "Wim Taymans <wim@fluendo.com>");

#define DEFAULT_DEVICE		"default"
#define DEFAULT_DEVICE_NAME	""

enum
{
  PROP_0,
  PROP_DEVICE,
  PROP_DEVICE_NAME
};

static void gst_alsasink_base_init (gpointer g_class);
static void gst_alsasink_class_init (GstAlsaSinkClass * klass);
static void gst_alsasink_init (GstAlsaSink * alsasink);
static void gst_alsasink_dispose (GObject * object);
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);

static gboolean gst_alsasink_open (GstAudioSink * asink);
static gboolean gst_alsasink_prepare (GstAudioSink * asink,
    GstRingBufferSpec * spec);
static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
static gboolean gst_alsasink_close (GstAudioSink * asink);
static guint gst_alsasink_write (GstAudioSink * asink, gpointer data,
    guint length);
static guint gst_alsasink_delay (GstAudioSink * asink);
static void gst_alsasink_reset (GstAudioSink * asink);

/* AlsaSink signals and args */
enum
{
  LAST_SIGNAL
};

#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ALSA_SINK_FACTORY_ENDIANNESS	"LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ALSA_SINK_FACTORY_ENDIANNESS	"BIG_ENDIAN, LITTLE_ENDIAN"
#endif

static GstStaticPadTemplate alsasink_sink_factory =
    GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-raw-int, "
        "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
        "signed = (boolean) { TRUE, FALSE }, "
        "width = (int) 32, "
        "depth = (int) 32, "
        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 8 ]; "
        "audio/x-raw-int, "
        "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
        "signed = (boolean) { TRUE, FALSE }, "
        "width = (int) 16, "
        "depth = (int) 16, "
        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 8 ]; "
        "audio/x-raw-int, "
        "signed = (boolean) { TRUE, FALSE }, "
        "width = (int) 8, "
        "depth = (int) 8, "
        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 8 ]")
    );

static GstElementClass *parent_class = NULL;

GType
gst_alsasink_get_type (void)
{
  static GType alsasink_type = 0;

  if (!alsasink_type) {
    static const GTypeInfo alsasink_info = {
      sizeof (GstAlsaSinkClass),
      gst_alsasink_base_init,
      NULL,
      (GClassInitFunc) gst_alsasink_class_init,
      NULL,
      NULL,
      sizeof (GstAlsaSink),
      0,
      (GInstanceInitFunc) gst_alsasink_init,
    };

    alsasink_type =
        g_type_register_static (GST_TYPE_AUDIO_SINK, "GstAlsaSink",
        &alsasink_info, 0);
  }

  return alsasink_type;
}

static void
gst_alsasink_dispose (GObject * object)
{
  G_OBJECT_CLASS (parent_class)->dispose (object);
}

static void
gst_alsasink_finalise (GObject * object)
{
  GstAlsaSink *sink = GST_ALSA_SINK (object);

  g_free (sink->device);
  g_mutex_free (sink->alsa_lock);

  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static void
gst_alsasink_base_init (gpointer g_class)
{
  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

  gst_element_class_set_details (element_class, &gst_alsasink_details);

  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&alsasink_sink_factory));
}
static void
gst_alsasink_class_init (GstAlsaSinkClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstBaseSinkClass *gstbasesink_class;
  GstBaseAudioSinkClass *gstbaseaudiosink_class;
  GstAudioSinkClass *gstaudiosink_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;
  gstbasesink_class = (GstBaseSinkClass *) klass;
  gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
  gstaudiosink_class = (GstAudioSinkClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_alsasink_dispose);
  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise);
  gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property);
  gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property);

  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);

  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_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));
}

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:
      if (sink->device)
        g_free (sink->device);
      sink->device = g_strdup (g_value_get_string (value));
      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:
      if (sink->handle) {
        snd_pcm_info_t *info;

        snd_pcm_info_malloc (&info);
        snd_pcm_info (sink->handle, info);
        g_value_set_string (value, snd_pcm_info_get_name (info));
        snd_pcm_info_free (info);
      } else {
        g_value_set_string (value, NULL);
      }
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static snd_output_t *output = NULL;

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;
  alsasink->alsa_lock = g_mutex_new ();

  snd_output_stdio_attach (&output, stdout, 0);
}

#define CHECK(call, error) \
G_STMT_START {                  \
if ((err = call) < 0)           \
  goto error;                   \
} G_STMT_END;

/* we don't have channel mappings for more than this many channels */
#define GST_ALSA_MAX_CHANNELS 8

static GstStructure *
get_channel_free_structure (const GstStructure * in_structure)
{
  GstStructure *s = gst_structure_copy (in_structure);

  gst_structure_remove_field (s, "channels");
  return s;
}

static void
caps_add_channel_configuration (GstCaps * caps,
    const GstStructure * in_structure, gint min_channels, gint max_channels)
{
  GstAudioChannelPosition pos[8] = {
    GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
    GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
    GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
    GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
    GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
    GST_AUDIO_CHANNEL_POSITION_LFE,
    GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
    GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
  };
  GstStructure *s = NULL;
  gint c;

  if (min_channels == max_channels) {
    s = get_channel_free_structure (in_structure);
    gst_structure_set (s, "channels", G_TYPE_INT, max_channels, NULL);
    gst_caps_append_structure (caps, s);
    return;
  }

  g_assert (min_channels >= 1);

  /* mono and stereo don't need channel configurations */
  if (min_channels == 2) {
    s = get_channel_free_structure (in_structure);
    gst_structure_set (s, "channels", G_TYPE_INT, 2, NULL);
    gst_caps_append_structure (caps, s);
  } else if (min_channels == 1 && max_channels >= 2) {
    s = get_channel_free_structure (in_structure);
    gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
    gst_caps_append_structure (caps, s);
  }

  /* don't know whether to use 2.1 or 3.0 here - but I suspect
   * alsa might work around that/fix it somehow. Can we tell alsa
   * what our channel layout is like? */
  if (max_channels >= 3) {
    GstAudioChannelPosition pos_21[3] = {
      GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
      GST_AUDIO_CHANNEL_POSITION_LFE
    };

    s = get_channel_free_structure (in_structure);
    gst_structure_set (s, "channels", G_TYPE_INT, 3, NULL);
    gst_audio_set_channel_positions (s, pos_21);
    gst_caps_append_structure (caps, s);
  }

  /* everything else (4, 6, 8 channels) needs a channel layout */
  for (c = 4; c < 8; c += 2) {
    if (max_channels >= c) {
      s = get_channel_free_structure (in_structure);
      gst_structure_set (s, "channels", G_TYPE_INT, c, NULL);
      gst_audio_set_channel_positions (s, pos);
      gst_caps_append_structure (caps, s);
    }
  }
}

static GstCaps *
gst_alsasink_getcaps (GstBaseSink * bsink)
{
  snd_pcm_format_mask_t *mask;
  snd_pcm_hw_params_t *hw_params;
  GstElementClass *element_class;
  GstPadTemplate *pad_template;
  GstAlsaSink *sink = GST_ALSA_SINK (bsink);
  GstCaps *tmpl_caps;
  GstCaps *caps = NULL;
  GstStructure *s;
  guint min, max;
  gint i, err, width;
  gint min_channels, max_channels;
  guint bits = 0;
  static const int audio_fmts[] = {
    SND_PCM_FORMAT_U8, SND_PCM_FORMAT_S8,
    SND_PCM_FORMAT_S16, SND_PCM_FORMAT_U16,
    /*SND_PCM_FORMAT_S24, SND_PCM_FORMAT_U24, */
    SND_PCM_FORMAT_S32, SND_PCM_FORMAT_U32
  };
  static const guint audio_bits[] = {
    8, 8, 16, 16, 32, 32
  };

  if (sink->handle == NULL) {
    GST_DEBUG_OBJECT (sink, "device not open, using template caps");
    return NULL;                /* base class will get template caps for us */
  }

  if (sink->cached_caps) {
    GST_DEBUG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT,
        sink->cached_caps);
    return gst_caps_ref (sink->cached_caps);
  }

  snd_pcm_hw_params_alloca (&hw_params);
  CHECK (snd_pcm_hw_params_any (sink->handle, hw_params), error);

  GST_LOG_OBJECT (sink, "probing channels ...");
  CHECK (snd_pcm_hw_params_get_channels_min (hw_params, &min), min_chan_error);
  CHECK (snd_pcm_hw_params_get_channels_max (hw_params, &max), max_chan_error);

  min_channels = min;
  max_channels = max;

  if (min_channels < 0) {       /* hmm? min and max are unsigned */
    min_channels = 1;
    max_channels = GST_ALSA_MAX_CHANNELS;
  } else if (max_channels < 0) {        /* hmm? min and max are unsigned */
    max_channels = GST_ALSA_MAX_CHANNELS;
  }

  if (min_channels > max_channels) {
    gint temp;

    GST_WARNING_OBJECT (sink, "minimum channels > maximum channels (%d > %d), "
        "please fix your soundcard drivers", min, max);
    temp = min_channels;
    min_channels = max_channels;
    max_channels = temp;
  }

  min_channels = MAX (min_channels, 1);
  max_channels = MIN (GST_ALSA_MAX_CHANNELS, max_channels);

  GST_LOG_OBJECT (sink, "Min. channels = %d (%d)", min_channels, min);
  GST_LOG_OBJECT (sink, "Max. channels = %d (%d)", max_channels, max);

  snd_pcm_format_mask_alloca (&mask);
  snd_pcm_hw_params_get_format_mask (hw_params, mask);
  for (i = 0; i < 6; i++) {
    if (snd_pcm_format_mask_test (mask, audio_fmts[i])) {
      bits |= audio_bits[i];
    }
  }
  GST_LOG_OBJECT (sink, "Bits = 0x%08x", bits);

  /* fill caps according to capabilities gathered above */

  element_class = GST_ELEMENT_GET_CLASS (sink);
  pad_template = gst_element_class_get_pad_template (element_class, "sink");

  g_return_val_if_fail (pad_template != NULL, NULL);

  tmpl_caps = gst_pad_template_get_caps (pad_template);

  caps = gst_caps_new_empty ();

  for (i = 0; i < gst_caps_get_size (tmpl_caps); ++i) {
    s = gst_caps_get_structure (tmpl_caps, i);
    gst_structure_get_int (s, "width", &width);
    /* TODO: filter signed/unsigned */
    if (bits & width) {
      caps_add_channel_configuration (caps, s, min_channels, max_channels);
    } else {
      GST_LOG_OBJECT (sink, "width = %d unsupported", width);
    }
  }

  sink->cached_caps = gst_caps_ref (caps);

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

  return caps;

error:
  {
    GST_ERROR_OBJECT (sink, "failed to query alsasink formats: %s",
        snd_strerror (err));
    return NULL;
  }
min_chan_error:
  {
    GST_ERROR_OBJECT (sink, "failed to query minimum channel count: %s",
        snd_strerror (err));
    return NULL;
  }
max_chan_error:
  {
    GST_ERROR_OBJECT (sink, "failed to query maximum channel count: %s",
        snd_strerror (err));
    return NULL;
  }
}

static int
set_hwparams (GstAlsaSink * alsa)
{
  guint rrate;
  gint err, dir;
  snd_pcm_hw_params_t *params;

  snd_pcm_hw_params_alloca (&params);

  GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %d)",
      alsa->channels, alsa->rate, alsa->format);

  /* 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 */
  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, 0),
      no_rate);
  if (rrate != alsa->rate)
    goto rate_match;

  if (alsa->buffer_time != -1) {
    /* set the buffer time */
    CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
            &alsa->buffer_time, &dir), buffer_time);
  }
  if (alsa->period_time != -1) {
    /* set the period time */
    CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
            &alsa->period_time, &dir), period_time);
  }

  /* write the parameters to device */
  CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);

  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, &dir),
      period_size);

  return 0;

  /* ERRORS */
no_config:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Broken configuration for playback: no configurations available: %s",
            snd_strerror (err)));
    return err;
  }
wrong_access:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Access type not available for playback: %s", snd_strerror (err)));
    return err;
  }
no_sample_format:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Sample format not available for playback: %s", snd_strerror (err)));
    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, (msg), (snd_strerror (err)));
    g_free (msg);
    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;
  }
rate_match:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
    return -EINVAL;
  }
buffer_time:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set buffer time %i for playback: %s",
            alsa->buffer_time, 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)));
    return err;
  }
period_time:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set period time %i for playback: %s", alsa->period_time,
            snd_strerror (err)));
    return err;
  }
period_size:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to get period size for playback: %s", snd_strerror (err)));
    return err;
  }
set_hw_params:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set hw params for playback: %s", snd_strerror (err)));
    return err;
  }
}

static int
set_swparams (GstAlsaSink * alsa)
{
  int err;
  snd_pcm_sw_params_t *params;

  snd_pcm_sw_params_alloca (&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);
  /* align all transfers to 1 sample */
  CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);

  /* write the parameters to the playback device */
  CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);

  return 0;

  /* ERRORS */
no_config:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to determine current swparams for playback: %s",
            snd_strerror (err)));
    return err;
  }
start_threshold:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set start threshold mode for playback: %s",
            snd_strerror (err)));
    return err;
  }
set_avail:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set avail min for playback: %s", snd_strerror (err)));
    return err;
  }
set_align:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set transfer align for playback: %s", snd_strerror (err)));
    return err;
  }
set_sw_params:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set sw params for playback: %s", snd_strerror (err)));
    return err;
  }
}

static gboolean
alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
{
  switch (spec->type) {
    case GST_BUFTYPE_LINEAR:
      GST_DEBUG_OBJECT (alsa,
          "Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth,
          spec->width, spec->sign, spec->bigend);

      alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
          spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
      break;
    case GST_BUFTYPE_FLOAT:
      switch (spec->format) {
        case GST_FLOAT32_LE:
          alsa->format = SND_PCM_FORMAT_FLOAT_LE;
          break;
        case GST_FLOAT32_BE:
          alsa->format = SND_PCM_FORMAT_FLOAT_BE;
          break;
        case GST_FLOAT64_LE:
          alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
          break;
        case GST_FLOAT64_BE:
          alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
          break;
        default:
          goto error;
      }
      break;
    case GST_BUFTYPE_A_LAW:
      alsa->format = SND_PCM_FORMAT_A_LAW;
      break;
    case GST_BUFTYPE_MU_LAW:
      alsa->format = SND_PCM_FORMAT_MU_LAW;
      break;
    default:
      goto error;

  }
  alsa->rate = spec->rate;
  alsa->channels = spec->channels;
  alsa->buffer_time = spec->buffer_time;
  alsa->period_time = spec->latency_time;
  alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;

  return TRUE;

  /* ERRORS */
error:
  {
    return FALSE;
  }
}

static gboolean
gst_alsasink_open (GstAudioSink * asink)
{
  GstAlsaSink *alsa;
  gint err;

  alsa = GST_ALSA_SINK (asink);

  CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
          SND_PCM_NONBLOCK), open_error);

  return TRUE;

  /* ERRORS */
open_error:
  {
    if (err == -EBUSY) {
      GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, (NULL), ("Device is busy"));
    } else {
      GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
          (NULL), ("Playback open error: %s", snd_strerror (err)));
    }
    return FALSE;
  }
}

static gboolean
gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
  GstAlsaSink *alsa;
  gint err;

  alsa = GST_ALSA_SINK (asink);

  if (!alsasink_parse_spec (alsa, spec))
    goto spec_parse;

  CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);

  CHECK (set_hwparams (alsa), hw_params_failed);
  CHECK (set_swparams (alsa), sw_params_failed);

  alsa->bytes_per_sample = spec->bytes_per_sample;
  spec->segsize = alsa->period_size * spec->bytes_per_sample;
  spec->segtotal = alsa->buffer_size / alsa->period_size;
  spec->silence_sample[0] = 0;
  spec->silence_sample[1] = 0;
  spec->silence_sample[2] = 0;
  spec->silence_sample[3] = 0;

  return TRUE;

  /* ERRORS */
spec_parse:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Error parsing spec"));
    return FALSE;
  }
non_block:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Could not set device to blocking: %s", snd_strerror (err)));
    return FALSE;
  }
hw_params_failed:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Setting of hwparams failed: %s", snd_strerror (err)));
    return FALSE;
  }
sw_params_failed:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Setting of swparams failed: %s", snd_strerror (err)));
    return FALSE;
  }
}

static gboolean
gst_alsasink_unprepare (GstAudioSink * asink)
{
  GstAlsaSink *alsa;
  gint err;

  alsa = GST_ALSA_SINK (asink);

  CHECK (snd_pcm_drop (alsa->handle), drop);

  CHECK (snd_pcm_hw_free (alsa->handle), hw_free);

  CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block);

  return TRUE;

  /* ERRORS */
drop:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Could not drop samples: %s", snd_strerror (err)));
    return FALSE;
  }
hw_free:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Could not free hw params: %s", snd_strerror (err)));
    return FALSE;
  }
non_block:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Could not set device to nonblocking: %s", snd_strerror (err)));
    return FALSE;
  }
}

static gboolean
gst_alsasink_close (GstAudioSink * asink)
{
  GstAlsaSink *alsa = GST_ALSA_SINK (asink);
  gint err;

  CHECK (snd_pcm_close (alsa->handle), close_error);
  alsa->handle = NULL;
  gst_caps_replace (&alsa->cached_caps, NULL);

  return TRUE;

  /* ERRORS */
close_error:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, CLOSE, (NULL),
        ("Playback close error: %s", snd_strerror (err)));
    return FALSE;
  }
}


/*
 *   Underrun and suspend recovery
 */
static gint
xrun_recovery (snd_pcm_t * handle, gint err)
{
  GST_DEBUG ("xrun recovery %d", err);

  if (err == -EPIPE) {          /* under-run */
    err = snd_pcm_prepare (handle);
    if (err < 0)
      GST_WARNING ("Can't recovery from underrun, prepare failed: %s",
          snd_strerror (err));
    return 0;
  } else if (err == -ESTRPIPE) {
    while ((err = snd_pcm_resume (handle)) == -EAGAIN)
      g_usleep (100);           /* wait until the suspend flag is released */

    if (err < 0) {
      err = snd_pcm_prepare (handle);
      if (err < 0)
        GST_WARNING ("Can't recovery from suspend, prepare failed: %s",
            snd_strerror (err));
    }
    return 0;
  }
  return err;
}

static guint
gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
{
  GstAlsaSink *alsa;
  gint err;
  gint cptr;
  gint16 *ptr;

  alsa = GST_ALSA_SINK (asink);

  cptr = length / alsa->bytes_per_sample;
  ptr = data;

  GST_ALSA_LOCK (asink);
  while (cptr > 0) {
    err = snd_pcm_writei (alsa->handle, ptr, cptr);

    GST_DEBUG_OBJECT (asink, "written %d result %d", cptr, err);
    if (err < 0) {
      GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
      if (err == -EAGAIN) {
        continue;
      } else if (xrun_recovery (alsa->handle, err) < 0) {
        goto write_error;
      }
      continue;
    }

    ptr += err * alsa->channels;
    cptr -= err;
  }
  GST_ALSA_UNLOCK (asink);

  return length - cptr;

write_error:
  {
    GST_ALSA_UNLOCK (asink);
    return length;              /* skip one period */
  }
}

static guint
gst_alsasink_delay (GstAudioSink * asink)
{
  GstAlsaSink *alsa;
  snd_pcm_sframes_t delay;

  alsa = GST_ALSA_SINK (asink);

  snd_pcm_delay (alsa->handle, &delay);

  return delay;
}

static void
gst_alsasink_reset (GstAudioSink * asink)
{
  GstAlsaSink *alsa;
  gint err;

  alsa = GST_ALSA_SINK (asink);

  GST_ALSA_LOCK (asink);
  GST_DEBUG_OBJECT (alsa, "drop");
  CHECK (snd_pcm_drop (alsa->handle), drop_error);
  GST_DEBUG_OBJECT (alsa, "prepare");
  CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
  GST_DEBUG_OBJECT (alsa, "reset done");
  GST_ALSA_UNLOCK (asink);

  return;

  /* ERRORS */
drop_error:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS,
        ("alsa-reset: pcm drop error: %s", snd_strerror (err)), (NULL));
    GST_ALSA_UNLOCK (asink);
    return;
  }
prepare_error:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS,
        ("alsa-reset: pcm prepare error: %s", snd_strerror (err)), (NULL));
    GST_ALSA_UNLOCK (asink);
    return;
  }
}