gstreamer/ext/artsd/gstartsdsink.c

/* GStreamer
 * Copyright (C) <2001> Richard Boulton <richard-gst@tartarus.org>
 *
 * Based on example.c:
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 *
 * 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.
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstartsdsink.h"
#include <gst/audio/audio.h>

/* Signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  ARG_0,
  ARG_MUTE,
  ARG_NAME
};

static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
    );

static void gst_artsdsink_base_init (gpointer g_class);
static void gst_artsdsink_class_init (GstArtsdsinkClass * klass);
static void gst_artsdsink_init (GstArtsdsink * artsdsink);

static gboolean gst_artsdsink_open_audio (GstArtsdsink * sink);
static void gst_artsdsink_close_audio (GstArtsdsink * sink);
static GstStateChangeReturn gst_artsdsink_change_state (GstElement * element,
    GstStateChange transition);
static gboolean gst_artsdsink_sync_parms (GstArtsdsink * artsdsink);
static GstPadLinkReturn gst_artsdsink_link (GstPad * pad, const GstCaps * caps);
static void gst_artsdsink_chain (GstPad * pad, GstData * _data);

static void gst_artsdsink_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_artsdsink_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);

static GstElementClass *parent_class = NULL;

/*static guint gst_artsdsink_signals[LAST_SIGNAL] = { 0 }; */

GType
gst_artsdsink_get_type (void)
{
  static GType artsdsink_type = 0;

  if (!artsdsink_type) {
    static const GTypeInfo artsdsink_info = {
      sizeof (GstArtsdsinkClass),
      gst_artsdsink_base_init,
      NULL,
      (GClassInitFunc) gst_artsdsink_class_init,
      NULL,
      NULL,
      sizeof (GstArtsdsink),
      0,
      (GInstanceInitFunc) gst_artsdsink_init,
    };

    artsdsink_type =
        g_type_register_static (GST_TYPE_ELEMENT, "GstArtsdsink",
        &artsdsink_info, 0);
  }
  return artsdsink_type;
}

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

  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&sink_factory));
  gst_element_class_set_details_simple (element_class, "aRtsd audio sink",
      "Sink/Audio",
      "Plays audio to an aRts server",
      "Richard Boulton <richard-gst@tartarus.org>");
}

static void
gst_artsdsink_class_init (GstArtsdsinkClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MUTE, g_param_spec_boolean ("mute", "mute", "mute", TRUE, G_PARAM_READWRITE));   /* CHECKME */

  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_NAME, g_param_spec_string ("name", "name", "name", NULL, G_PARAM_READWRITE));    /* CHECKME */

  gobject_class->set_property = gst_artsdsink_set_property;
  gobject_class->get_property = gst_artsdsink_get_property;

  gstelement_class->change_state = gst_artsdsink_change_state;
}

static void
gst_artsdsink_init (GstArtsdsink * artsdsink)
{
  artsdsink->sinkpad =
      gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
          (artsdsink), "sink"), "sink");
  gst_element_add_pad (GST_ELEMENT (artsdsink), artsdsink->sinkpad);
  gst_pad_set_chain_function (artsdsink->sinkpad, gst_artsdsink_chain);
  gst_pad_set_link_function (artsdsink->sinkpad, gst_artsdsink_link);

  artsdsink->connected = FALSE;
  artsdsink->mute = FALSE;
  artsdsink->connect_name = NULL;
}

static gboolean
gst_artsdsink_sync_parms (GstArtsdsink * artsdsink)
{
  g_return_val_if_fail (artsdsink != NULL, FALSE);
  g_return_val_if_fail (GST_IS_ARTSDSINK (artsdsink), FALSE);

  if (!artsdsink->connected)
    return TRUE;

  /* Need to set stream to use new parameters: only way to do this is to reopen. */
  gst_artsdsink_close_audio (artsdsink);
  return gst_artsdsink_open_audio (artsdsink);
}

static GstPadLinkReturn
gst_artsdsink_link (GstPad * pad, const GstCaps * caps)
{
  GstArtsdsink *artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
  GstStructure *structure;

  structure = gst_caps_get_structure (caps, 0);
  gst_structure_get_int (structure, "rate", &artsdsink->frequency);
  gst_structure_get_int (structure, "depth", &artsdsink->depth);
  gst_structure_get_int (structure, "signed", &artsdsink->signd);
  gst_structure_get_int (structure, "channels", &artsdsink->channels);

  if (gst_artsdsink_sync_parms (artsdsink))
    return GST_PAD_LINK_OK;

  return GST_PAD_LINK_REFUSED;
}

static void
gst_artsdsink_chain (GstPad * pad, GstData * _data)
{
  GstBuffer *buf = GST_BUFFER (_data);
  GstArtsdsink *artsdsink;

  g_return_if_fail (pad != NULL);
  g_return_if_fail (GST_IS_PAD (pad));
  g_return_if_fail (buf != NULL);

  artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));

  if (GST_BUFFER_DATA (buf) != NULL) {
    gst_trace_add_entry (NULL, 0, GPOINTER_TO_INT (buf),
        "artsdsink: writing to server");
    if (!artsdsink->mute && artsdsink->connected) {
      int bytes;
      void *bufptr = GST_BUFFER_DATA (buf);
      int bufsize = GST_BUFFER_SIZE (buf);

      GST_DEBUG ("artsdsink: stream=%p data=%p size=%d",
          artsdsink->stream, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));

      do {
        bytes = arts_write (artsdsink->stream, bufptr, bufsize);
        if (bytes < 0) {
          fprintf (stderr, "arts_write error: %s\n", arts_error_text (bytes));
          gst_buffer_unref (buf);
          return;
        }
        bufptr += bytes;
        bufsize -= bytes;
      } while (bufsize > 0);
    }
  }
  gst_buffer_unref (buf);
}

static void
gst_artsdsink_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstArtsdsink *artsdsink;

  g_return_if_fail (GST_IS_ARTSDSINK (object));
  artsdsink = GST_ARTSDSINK (object);

  switch (prop_id) {
    case ARG_MUTE:
      artsdsink->mute = g_value_get_boolean (value);
      break;
    case ARG_NAME:
      if (artsdsink->connect_name != NULL)
        g_free (artsdsink->connect_name);
      if (g_value_get_string (value) == NULL)
        artsdsink->connect_name = NULL;
      else
        artsdsink->connect_name = g_strdup (g_value_get_string (value));
      break;
    default:
      break;
  }
}

static void
gst_artsdsink_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstArtsdsink *artsdsink;

  g_return_if_fail (GST_IS_ARTSDSINK (object));
  artsdsink = GST_ARTSDSINK (object);

  switch (prop_id) {
    case ARG_MUTE:
      g_value_set_boolean (value, artsdsink->mute);
      break;
    case ARG_NAME:
      g_value_set_string (value, artsdsink->connect_name);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static gboolean
plugin_init (GstPlugin * plugin)
{
  if (!gst_element_register (plugin, "artsdsink", GST_RANK_NONE,
          GST_TYPE_ARTSDSINK))
    return FALSE;

  return TRUE;
}

GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
    GST_VERSION_MINOR,
    "artsdsink",
    "Plays audio to an aRts server",
    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

     static gboolean gst_artsdsink_open_audio (GstArtsdsink * sink)
{
  const char connname[] = "gstreamer";
  int errcode;

  /* Name used by aRtsd for this connection. */
  if (sink->connect_name != NULL)
    connname = sink->connect_name;

  /* FIXME: this should only ever happen once per process. */
  /* Really, artsc needs to be made thread safe to fix this (and other related */
  /* problems). */
  errcode = arts_init ();
  if (errcode < 0) {
    fprintf (stderr, "arts_init error: %s\n", arts_error_text (errcode));
    return FALSE;
  }

  GST_DEBUG ("artsdsink: attempting to open connection to aRtsd server");
  sink->stream = arts_play_stream (sink->frequency, sink->depth,
      sink->channels, connname);
  /* FIXME: check connection */
  /*   GST_DEBUG ("artsdsink: can't open connection to aRtsd server"); */

  GST_OBJECT_FLAG_SET (sink, GST_ARTSDSINK_OPEN);
  sink->connected = TRUE;

  return TRUE;
}

static void
gst_artsdsink_close_audio (GstArtsdsink * sink)
{
  if (!sink->connected)
    return;

  arts_close_stream (sink->stream);
  arts_free ();
  GST_OBJECT_FLAG_UNSET (sink, GST_ARTSDSINK_OPEN);
  sink->connected = FALSE;

  g_print ("artsdsink: closed connection\n");
}

static GstStateChangeReturn
gst_artsdsink_change_state (GstElement * element, GstStateChange transition)
{
  g_return_val_if_fail (GST_IS_ARTSDSINK (element), FALSE);

  /* if going down into NULL state, close the stream if it's open */
  if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
    if (GST_OBJECT_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN))
      gst_artsdsink_close_audio (GST_ARTSDSINK (element));
    /* otherwise (READY or higher) we need to open the stream */
  } else {
    if (!GST_OBJECT_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN)) {
      if (!gst_artsdsink_open_audio (GST_ARTSDSINK (element)))
        return GST_STATE_CHANGE_FAILURE;
    }
  }

  if (GST_ELEMENT_CLASS (parent_class)->change_state)
    return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
  return GST_STATE_CHANGE_SUCCESS;
}