1 /* GStreamer audio filter base class

     2  * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>

     3  * Copyright (C) <2003> David Schleef <ds@schleef.org>

     4  * Copyright (C) <2007> Tim-Philipp Müller <tim centricular net>

     5  *

     6  * This library is free software; you can redistribute it and/or

     7  * modify it under the terms of the GNU Library General Public

     8  * License as published by the Free Software Foundation; either

     9  * version 2 of the License, or (at your option) any later version.

    10  *

    11  * This library is distributed in the hope that it will be useful,

    12  * but WITHOUT ANY WARRANTY; without even the implied warranty of

    13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

    14  * Library General Public License for more details.

    15  *

    16  * You should have received a copy of the GNU Library General Public

    17  * License along with this library; if not, write to the

    18  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,

    19  * Boston, MA 02111-1307, USA.

    20  */

    21 

    22 /**

    23  * SECTION:gstaudiofilter

    24  * @short_description: Base class for simple audio filters

    25  *

    26  * #GstAudioFilter is a #GstBaseTransform-derived base class for simple audio

    27  * filters, ie. those that output the same format that they get as input.

    28  *

    29  * #GstAudioFilter will parse the input format for you (with error checking)

    30  * before calling your setup function. Also, elements deriving from

    31  * #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from

    32  * their base_init function to easily configure the set of caps/formats that

    33  * the element is able to handle.

    34  *

    35  * Derived classes should override the GstAudioFilter::setup() and

    36  * GstBaseTransform::transform_ip() and/or GstBaseTransform::transform()

    37  * virtual functions in their class_init function.

    38  *

    39  * Since: 0.10.12

    40  *

    41  * Last reviewed on 2007-02-03 (0.10.11.1)

    42  */

    43 

    44 #ifdef HAVE_CONFIG_H

    45 #include "config.h"

    46 #endif

    47 

    48 #include "gstaudiofilter.h"

    49 

    50 #include <string.h>

    51 

    52 GST_DEBUG_CATEGORY_STATIC (audiofilter_dbg);

    53 #define GST_CAT_DEFAULT audiofilter_dbg

    54 

    55 static void gst_audio_filter_class_init (gpointer g_class, gpointer class_data);

    56 static void gst_audio_filter_init (GTypeInstance * instance, gpointer g_class);

    57 static GstStateChangeReturn gst_audio_filter_change_state (GstElement * element,

    58     GstStateChange transition);

    59 static gboolean gst_audio_filter_set_caps (GstBaseTransform * btrans,

    60     GstCaps * incaps, GstCaps * outcaps);

    61 static gboolean gst_audio_filter_get_unit_size (GstBaseTransform * btrans,

    62     GstCaps * caps, guint * size);

    63 

    64 static GstElementClass *parent_class = NULL;

    65 #ifdef __SYMBIAN32__

    66 EXPORT_C

    67 #endif

    68 

    69 

    70 GType

    71 gst_audio_filter_get_type (void)

    72 {

    73   static GType audio_filter_type = 0;

    74 

    75   if (!audio_filter_type) {

    76     const GTypeInfo audio_filter_info = {

    77       sizeof (GstAudioFilterClass),

    78       NULL,

    79       NULL,

    80       gst_audio_filter_class_init,

    81       NULL,

    82       NULL,

    83       sizeof (GstAudioFilter),

    84       0,

    85       gst_audio_filter_init,

    86     };

    87 

    88     GST_DEBUG_CATEGORY_INIT (audiofilter_dbg, "audiofilter", 0, "audiofilter");

    89 

    90     audio_filter_type = g_type_register_static (GST_TYPE_BASE_TRANSFORM,

    91         "GstAudioFilter", &audio_filter_info, G_TYPE_FLAG_ABSTRACT);

    92   }

    93   return audio_filter_type;

    94 }

    95 

    96 static void

    97 gst_audio_filter_class_init (gpointer klass, gpointer class_data)

    98 {

    99   GstBaseTransformClass *basetrans_class;

   100   GstElementClass *gstelement_class;

   101 

   102   parent_class = g_type_class_peek_parent (klass);

   103 

   104   gstelement_class = (GstElementClass *) klass;

   105   basetrans_class = (GstBaseTransformClass *) klass;

   106 

   107   gstelement_class->change_state =

   108       GST_DEBUG_FUNCPTR (gst_audio_filter_change_state);

   109   basetrans_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_filter_set_caps);

   110   basetrans_class->get_unit_size =

   111       GST_DEBUG_FUNCPTR (gst_audio_filter_get_unit_size);

   112 

   113   /* FIXME: Ref the GstRingerBuffer class to get it's debug category

   114    * initialized. gst_ring_buffer_parse_caps () which we use later

   115    * uses this debug category.

   116    */

   117   g_type_class_ref (GST_TYPE_RING_BUFFER);

   118 }

   119 

   120 static void

   121 gst_audio_filter_init (GTypeInstance * instance, gpointer g_class)

   122 {

   123   /* nothing to do here */

   124 }

   125 

   126 /* we override the state change vfunc here instead of GstBaseTransform's stop

   127  * vfunc, so GstAudioFilter-derived elements can override ::stop() for their

   128  * own purposes without having to worry about chaining up */

   129 static GstStateChangeReturn

   130 gst_audio_filter_change_state (GstElement * element, GstStateChange transition)

   131 {

   132   GstStateChangeReturn ret;

   133   GstAudioFilter *filter;

   134 

   135   filter = GST_AUDIO_FILTER (element);

   136 

   137   switch (transition) {

   138     case GST_STATE_CHANGE_NULL_TO_READY:

   139       memset (&filter->format, 0, sizeof (GstRingBufferSpec));

   140       /* to make gst_buffer_spec_parse_caps() happy */

   141       filter->format.latency_time = GST_SECOND;

   142       break;

   143     default:

   144       break;

   145   }

   146 

   147   ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

   148   if (ret == GST_STATE_CHANGE_FAILURE)

   149     return ret;

   150 

   151   switch (transition) {

   152     case GST_STATE_CHANGE_PAUSED_TO_READY:

   153     case GST_STATE_CHANGE_READY_TO_NULL:

   154       gst_caps_replace (&filter->format.caps, NULL);

   155       break;

   156     default:

   157       break;

   158   }

   159 

   160   return ret;

   161 }

   162 

   163 static gboolean

   164 gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,

   165     GstCaps * outcaps)

   166 {

   167   GstAudioFilterClass *klass;

   168   GstAudioFilter *filter;

   169   gboolean ret = TRUE;

   170 

   171   g_assert (gst_caps_is_equal (incaps, outcaps));

   172 

   173   filter = GST_AUDIO_FILTER (btrans);

   174 

   175   GST_LOG_OBJECT (filter, "caps: %" GST_PTR_FORMAT, incaps);

   176 

   177   if (!gst_ring_buffer_parse_caps (&filter->format, incaps)) {

   178     GST_WARNING_OBJECT (filter, "couldn't parse %" GST_PTR_FORMAT, incaps);

   179     return FALSE;

   180   }

   181 

   182   klass = GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (filter));

   183 

   184   if (klass->setup)

   185     ret = klass->setup (filter, &filter->format);

   186 

   187   return ret;

   188 }

   189 

   190 static gboolean

   191 gst_audio_filter_get_unit_size (GstBaseTransform * btrans, GstCaps * caps,

   192     guint * size)

   193 {

   194   GstStructure *structure;

   195   gboolean ret = TRUE;

   196   gint width, channels;

   197 

   198   structure = gst_caps_get_structure (caps, 0);

   199 

   200   ret &= gst_structure_get_int (structure, "width", &width);

   201   ret &= gst_structure_get_int (structure, "channels", &channels);

   202 

   203   if (ret)

   204     *size = (width / 8) * channels;

   205 

   206   return ret;

   207 }

   208 

   209 /**

   210  * gst_audio_filter_class_add_pad_templates:

   211  * @klass: an #GstAudioFilterClass

   212  * @allowed_caps: what formats the filter can handle, as #GstCaps

   213  *

   214  * Convenience function to add pad templates to this element class, with

   215  * @allowed_caps as the caps that can be handled.

   216  *

   217  * This function is usually used from within a GObject base_init function.

   218  *

   219  * Since: 0.10.12

   220  */

   221 #ifdef __SYMBIAN32__

   222 EXPORT_C

   223 #endif

   224 

   225 void

   226 gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,

   227     const GstCaps * allowed_caps)

   228 {

   229   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

   230 

   231   g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));

   232   g_return_if_fail (allowed_caps != NULL);

   233   g_return_if_fail (GST_IS_CAPS (allowed_caps));

   234 

   235   gst_element_class_add_pad_template (element_class,

   236       gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,

   237           gst_caps_copy (allowed_caps)));

   238 

   239   gst_element_class_add_pad_template (element_class,

   240       gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,

   241           gst_caps_copy (allowed_caps)));

   242 }