gstreamer/gst-libs/gst/audio/gstaudiofilter.c

/* GStreamer
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 * Copyright (C) <2003> David Schleef <ds@schleef.org>
 *
 * 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

/*#define DEBUG_ENABLED */
#include "gstaudiofilter.h"

#include <string.h>

static const GstElementDetails audio_filter_details =
GST_ELEMENT_DETAILS ("Audio filter base class",
    "Filter/Effect/Audio",
    "Filters audio",
    "David Schleef <ds@schleef.org>");

/* GstAudioFilter signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  ARG_0,
  ARG_METHOD
      /* FILL ME */
};

static void gst_audio_filter_base_init (gpointer g_class);
static void gst_audio_filter_class_init (gpointer g_class, gpointer class_data);
static void gst_audio_filter_init (GTypeInstance * instance, gpointer g_class);

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

static GstFlowReturn gst_audio_filter_chain (GstPad * pad, GstBuffer * buffer);
GstCaps *gst_audio_filter_class_get_capslist (GstAudioFilterClass * klass);

static GstElementClass *parent_class = NULL;

GType
gst_audio_filter_get_type (void)
{
  static GType audio_filter_type = 0;

  if (!audio_filter_type) {
    static const GTypeInfo audio_filter_info = {
      sizeof (GstAudioFilterClass),
      gst_audio_filter_base_init,
      NULL,
      gst_audio_filter_class_init,
      NULL,
      NULL,
      sizeof (GstAudioFilter),
      0,
      gst_audio_filter_init,
    };

    audio_filter_type = g_type_register_static (GST_TYPE_ELEMENT,
        "GstAudioFilter", &audio_filter_info, G_TYPE_FLAG_ABSTRACT);
  }
  return audio_filter_type;
}

static void
gst_audio_filter_base_init (gpointer g_class)
{
  GstAudioFilterClass *klass = (GstAudioFilterClass *) g_class;
  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  gst_element_class_set_details (element_class, &audio_filter_details);
}

static void
gst_audio_filter_class_init (gpointer g_class, gpointer class_data)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstAudioFilterClass *klass;

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

  parent_class = g_type_class_peek_parent (klass);

  gobject_class->set_property = gst_audio_filter_set_property;
  gobject_class->get_property = gst_audio_filter_get_property;
}

static GstPadLinkReturn
gst_audio_filter_link (GstPad * pad, GstPad * peer)
{
  GstAudioFilter *audiofilter;

  //GstPadLinkReturn ret;
  //GstPadLinkReturn link_ret;
  //GstStructure *structure;
  GstAudioFilterClass *audio_filter_class;

  GST_DEBUG ("gst_audio_filter_link");
  audiofilter = GST_AUDIO_FILTER (gst_pad_get_parent (pad));
  audio_filter_class =
      GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (audiofilter));

#if 0
  ret = GST_PAD_LINK_DELAYED;   /* intialise with dummy value */
  if (pad == audiofilter->srcpad) {
    link_ret = gst_pad_try_set_caps (audiofilter->sinkpad, caps);
  } else {
    link_ret = gst_pad_try_set_caps (audiofilter->srcpad, caps);
  }

  if (GST_PAD_LINK_FAILED (link_ret)) {
    gst_object_unref (audiofilter);
    return link_ret;
  }

  structure = gst_caps_get_structure (caps, 0);

  if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
    ret = gst_structure_get_int (structure, "depth", &audiofilter->depth);
    ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
  } else if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float")
      == 0) {
    ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
  } else {
    g_assert_not_reached ();
  }
  ret &= gst_structure_get_int (structure, "rate", &audiofilter->rate);
  ret &= gst_structure_get_int (structure, "channels", &audiofilter->channels);

  if (!ret) {
    gst_object_unref (audiofilter);
    return GST_PAD_LINK_REFUSED;
  }

  audiofilter->bytes_per_sample = (audiofilter->width / 8) *
      audiofilter->channels;

  if (audio_filter_class->setup)
    (audio_filter_class->setup) (audiofilter);
#endif

  gst_object_unref (audiofilter);
  return GST_PAD_LINK_OK;
}

static void
gst_audio_filter_init (GTypeInstance * instance, gpointer g_class)
{
  GstAudioFilter *audiofilter = GST_AUDIO_FILTER (instance);
  GstPadTemplate *pad_template;

  GST_DEBUG ("gst_audio_filter_init");

  pad_template =
      gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
  g_return_if_fail (pad_template != NULL);
  audiofilter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
  gst_element_add_pad (GST_ELEMENT (audiofilter), audiofilter->sinkpad);
  gst_pad_set_chain_function (audiofilter->sinkpad, gst_audio_filter_chain);
  gst_pad_set_link_function (audiofilter->sinkpad, gst_audio_filter_link);
  //gst_pad_set_getcaps_function (audiofilter->sinkpad, gst_pad_proxy_getcaps);

  pad_template =
      gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
  g_return_if_fail (pad_template != NULL);
  audiofilter->srcpad = gst_pad_new_from_template (pad_template, "src");
  gst_element_add_pad (GST_ELEMENT (audiofilter), audiofilter->srcpad);
  gst_pad_set_link_function (audiofilter->srcpad, gst_audio_filter_link);
  //gst_pad_set_getcaps_function (audiofilter->srcpad, gst_pad_proxy_getcaps);

  audiofilter->inited = FALSE;
}

static GstFlowReturn
gst_audio_filter_chain (GstPad * pad, GstBuffer * buffer)
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *inbuf = GST_BUFFER (buffer);
  GstAudioFilter *audiofilter;
  GstBuffer *outbuf;
  GstAudioFilterClass *audio_filter_class;

  GST_DEBUG ("gst_audio_filter_chain");

  g_return_val_if_fail (pad != NULL, GST_FLOW_ERROR);
  g_return_val_if_fail (GST_IS_PAD (pad), GST_FLOW_ERROR);
  g_return_val_if_fail (inbuf != NULL, GST_FLOW_ERROR);

  audiofilter = GST_AUDIO_FILTER (gst_pad_get_parent (pad));
  /* g_return_if_fail (audiofilter->inited); */
  audio_filter_class =
      GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (audiofilter));

  GST_DEBUG ("gst_audio_filter_chain: got buffer of %d bytes in '%s'",
      GST_BUFFER_SIZE (inbuf), GST_OBJECT_NAME (audiofilter));

  if (audiofilter->passthru) {
    ret = gst_pad_push (audiofilter->srcpad, buffer);
    gst_object_unref (audiofilter);
    return ret;
  }

  audiofilter->size = GST_BUFFER_SIZE (inbuf);
  audiofilter->n_samples = audiofilter->size / audiofilter->bytes_per_sample;

  if (gst_buffer_is_writable (buffer)) {
    if (audio_filter_class->filter_inplace) {
      (audio_filter_class->filter_inplace) (audiofilter, inbuf);
      outbuf = inbuf;
    } else {
      outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE (inbuf));
      GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (inbuf);
      GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (inbuf);

      (audio_filter_class->filter) (audiofilter, outbuf, inbuf);
      gst_buffer_unref (inbuf);
    }
  } else {
    outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE (inbuf));
    GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (inbuf);
    GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (inbuf);
    if (audio_filter_class->filter) {
      (audio_filter_class->filter) (audiofilter, outbuf, inbuf);
    } else {
      memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf),
          GST_BUFFER_SIZE (inbuf));

      (audio_filter_class->filter_inplace) (audiofilter, outbuf);
    }
    gst_buffer_unref (inbuf);
  }

  ret = gst_pad_push (audiofilter->srcpad, outbuf);

  gst_object_unref (audiofilter);
  return ret;
}

static void
gst_audio_filter_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioFilter *src;

  g_return_if_fail (GST_IS_AUDIO_FILTER (object));
  src = GST_AUDIO_FILTER (object);

  GST_DEBUG ("gst_audio_filter_set_property");
  switch (prop_id) {
    default:
      break;
  }
}

static void
gst_audio_filter_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstAudioFilter *src;

  g_return_if_fail (GST_IS_AUDIO_FILTER (object));
  src = GST_AUDIO_FILTER (object);

  switch (prop_id) {
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

void
gst_audio_filter_class_add_pad_templates (GstAudioFilterClass *
    audio_filter_class, const GstCaps * caps)
{
  GstElementClass *element_class = GST_ELEMENT_CLASS (audio_filter_class);

  audio_filter_class->caps = gst_caps_copy (caps);

  gst_element_class_add_pad_template (element_class,
      gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
          gst_caps_copy (caps)));

  gst_element_class_add_pad_template (element_class,
      gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
          gst_caps_copy (caps)));
}