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a6e4222c70
Original commit message from CVS: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_class_init): Initialize the GstRingerBuffer class to get it's debug category initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug category and otherwise we get some g_critical(). Fixes bug #512334.
234 lines
7 KiB
C
234 lines
7 KiB
C
/* GStreamer audio filter base class
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2003> David Schleef <ds@schleef.org>
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* Copyright (C) <2007> Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstaudiofilter
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* @short_description: Base class for simple audio filters
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*
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* #GstAudioFilter is a #GstBaseTransform-derived base class for simple audio
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* filters, ie. those that output the same format that they get as input.
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*
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* #GstAudioFilter will parse the input format for you (with error checking)
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* before calling your setup function. Also, elements deriving from
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* #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
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* their base_init function to easily configure the set of caps/formats that
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* the element is able to handle.
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*
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* Derived classes should override the GstAudioFilter::setup() and
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* GstBaseTransform::transform_ip() and/or GstBaseTransform::transform()
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* virtual functions in their class_init function.
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*
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* Since: 0.10.12
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*
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* Last reviewed on 2007-02-03 (0.10.11.1)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstaudiofilter.h"
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (audiofilter_dbg);
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#define GST_CAT_DEFAULT audiofilter_dbg
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static void gst_audio_filter_class_init (gpointer g_class, gpointer class_data);
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static void gst_audio_filter_init (GTypeInstance * instance, gpointer g_class);
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static GstStateChangeReturn gst_audio_filter_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_audio_filter_set_caps (GstBaseTransform * btrans,
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GstCaps * incaps, GstCaps * outcaps);
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static gboolean gst_audio_filter_get_unit_size (GstBaseTransform * btrans,
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GstCaps * caps, guint * size);
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static GstElementClass *parent_class = NULL;
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GType
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gst_audio_filter_get_type (void)
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{
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static GType audio_filter_type = 0;
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if (!audio_filter_type) {
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const GTypeInfo audio_filter_info = {
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sizeof (GstAudioFilterClass),
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NULL,
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NULL,
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gst_audio_filter_class_init,
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NULL,
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NULL,
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sizeof (GstAudioFilter),
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0,
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gst_audio_filter_init,
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};
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GST_DEBUG_CATEGORY_INIT (audiofilter_dbg, "audiofilter", 0, "audiofilter");
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audio_filter_type = g_type_register_static (GST_TYPE_BASE_TRANSFORM,
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"GstAudioFilter", &audio_filter_info, G_TYPE_FLAG_ABSTRACT);
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}
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return audio_filter_type;
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}
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static void
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gst_audio_filter_class_init (gpointer klass, gpointer class_data)
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{
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GstBaseTransformClass *basetrans_class;
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GstElementClass *gstelement_class;
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parent_class = g_type_class_peek_parent (klass);
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gstelement_class = (GstElementClass *) klass;
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basetrans_class = (GstBaseTransformClass *) klass;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_audio_filter_change_state);
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basetrans_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_filter_set_caps);
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basetrans_class->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_audio_filter_get_unit_size);
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/* FIXME: Ref the GstRingerBuffer class to get it's debug category
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* initialized. gst_ring_buffer_parse_caps () which we use later
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* uses this debug category.
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*/
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g_type_class_ref (GST_TYPE_RING_BUFFER);
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}
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static void
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gst_audio_filter_init (GTypeInstance * instance, gpointer g_class)
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{
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/* nothing to do here */
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}
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/* we override the state change vfunc here instead of GstBaseTransform's stop
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* vfunc, so GstAudioFilter-derived elements can override ::stop() for their
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* own purposes without having to worry about chaining up */
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static GstStateChangeReturn
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gst_audio_filter_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstAudioFilter *filter;
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filter = GST_AUDIO_FILTER (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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memset (&filter->format, 0, sizeof (GstRingBufferSpec));
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/* to make gst_buffer_spec_parse_caps() happy */
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filter->format.latency_time = GST_SECOND;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret == GST_STATE_CHANGE_FAILURE)
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return ret;
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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case GST_STATE_CHANGE_READY_TO_NULL:
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gst_caps_replace (&filter->format.caps, NULL);
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break;
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default:
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break;
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}
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return ret;
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}
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static gboolean
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gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
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GstCaps * outcaps)
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{
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GstAudioFilterClass *klass;
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GstAudioFilter *filter;
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gboolean ret = TRUE;
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g_assert (gst_caps_is_equal (incaps, outcaps));
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filter = GST_AUDIO_FILTER (btrans);
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GST_LOG_OBJECT (filter, "caps: %" GST_PTR_FORMAT, incaps);
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if (!gst_ring_buffer_parse_caps (&filter->format, incaps)) {
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GST_WARNING_OBJECT (filter, "couldn't parse %" GST_PTR_FORMAT, incaps);
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return FALSE;
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}
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klass = GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (filter));
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if (klass->setup)
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ret = klass->setup (filter, &filter->format);
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return ret;
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}
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static gboolean
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gst_audio_filter_get_unit_size (GstBaseTransform * btrans, GstCaps * caps,
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guint * size)
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{
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GstStructure *structure;
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gboolean ret = TRUE;
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gint width, channels;
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structure = gst_caps_get_structure (caps, 0);
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ret &= gst_structure_get_int (structure, "width", &width);
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ret &= gst_structure_get_int (structure, "channels", &channels);
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if (ret)
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*size = (width / 8) * channels;
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return ret;
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}
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/**
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* gst_audio_filter_class_add_pad_templates:
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* @klass: an #GstAudioFilterClass
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* @allowed_caps: what formats the filter can handle, as #GstCaps
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*
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* Convenience function to add pad templates to this element class, with
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* @allowed_caps as the caps that can be handled.
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*
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* This function is usually used from within a GObject base_init function.
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*
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* Since: 0.10.12
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*/
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void
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gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,
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const GstCaps * allowed_caps)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));
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g_return_if_fail (allowed_caps != NULL);
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g_return_if_fail (GST_IS_CAPS (allowed_caps));
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gst_element_class_add_pad_template (element_class,
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gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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gst_caps_copy (allowed_caps)));
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gst_element_class_add_pad_template (element_class,
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gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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gst_caps_copy (allowed_caps)));
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}
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