Commit b63fff63 authored by Tim-Philipp Müller's avatar Tim-Philipp Müller
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Port GstAudioFilter to 0.10. This change technically breaks but seems...

Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...

Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes #403963 (and eventually also #403572).
Also document all of this a bit.
parent 7d78598f
2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes #403963 (and eventually also #403572).
Also document all of this a bit.
2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/utils/install-plugins.c:
......@@ -40,9 +40,6 @@ GST_IS_AUDIO_CLOCK_CLASS
<INCLUDE>gst/audio/gstaudiofilter.h</INCLUDE>
GstAudioFilter
GstAudioFilterClass
GstAudioFilterFilterFunc
GstAudioFilterInplaceFilterFunc
GstAudioFilterSetupFunc
gst_audio_filter_class_add_pad_templates
<SUBSECTION Standard>
GST_AUDIO_FILTER
......
/* GStreamer
/* GStreamer audio filter base class
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2003> David Schleef <ds@schleef.org>
* Copyright (C) <2007> Tim-Philipp Müller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
......@@ -18,46 +19,51 @@
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstaudiofilter
* @short_description: Base class for simple audio filters
*
* #GstAudioFilter is a #GstBaseTransform-derived base class for simple audio
* filters, ie. those that output the same format that they get as input.
*
* #GstAudioFilter will parse the input format for you (with error checking)
* before calling your setup function. Also, elements deriving from
* #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
* their base_init function to easily configure the set of caps/formats that
* the element is able to handle.
*
* Derived classes should override the GstAudioFilter::setup() and
* GstBaseTransform::transform_ip() and/or GstBaseTransform::transform()
* virtual functions in their class_init function.
*
* Since: 0.10.12
*
* Last reviewed on 2007-02-03 (0.10.11.1)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
/*#define DEBUG_ENABLED */
#include "gstaudiofilter.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audiofilter_dbg);
#define GST_CAT_DEFAULT audiofilter_dbg
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 gboolean gst_audio_filter_set_caps (GstBaseTransform * btrans,
GstCaps * incaps, GstCaps * outcaps);
static GstElementClass *parent_class = NULL;
......@@ -67,7 +73,7 @@ gst_audio_filter_get_type (void)
static GType audio_filter_type = 0;
if (!audio_filter_type) {
static const GTypeInfo audio_filter_info = {
const GTypeInfo audio_filter_info = {
sizeof (GstAudioFilterClass),
gst_audio_filter_base_init,
NULL,
......@@ -79,7 +85,9 @@ gst_audio_filter_get_type (void)
gst_audio_filter_init,
};
audio_filter_type = g_type_register_static (GST_TYPE_ELEMENT,
GST_DEBUG_CATEGORY_INIT (audiofilter_dbg, "audiofilter", 0, "audiofilter");
audio_filter_type = g_type_register_static (GST_TYPE_BASE_TRANSFORM,
"GstAudioFilter", &audio_filter_info, G_TYPE_FLAG_ABSTRACT);
}
return audio_filter_type;
......@@ -95,218 +103,80 @@ gst_audio_filter_base_init (gpointer g_class)
}
static void
gst_audio_filter_class_init (gpointer g_class, gpointer class_data)
gst_audio_filter_class_init (gpointer klass, gpointer class_data)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioFilterClass *klass;
klass = (GstAudioFilterClass *) g_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
GstBaseTransformClass *basetrans_class;
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;
basetrans_class = (GstBaseTransformClass *) klass;
if (audio_filter_class->setup)
(audio_filter_class->setup) (audiofilter);
#endif
gst_object_unref (audiofilter);
return GST_PAD_LINK_OK;
basetrans_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_filter_set_caps);
}
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;
GstAudioFilter *filter = GST_AUDIO_FILTER (instance);
/* to make gst_buffer_spec_parse_caps() happy, not used in our case */
filter->format.latency_time = GST_SECOND;
}
static GstFlowReturn
gst_audio_filter_chain (GstPad * pad, GstBuffer * buffer)
static gboolean
gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
GstCaps * outcaps)
{
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;
}
GstAudioFilterClass *klass;
GstAudioFilter *filter;
gboolean ret = TRUE;
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);
}
g_assert (gst_caps_is_equal (incaps, outcaps));
ret = gst_pad_push (audiofilter->srcpad, outbuf);
filter = GST_AUDIO_FILTER (btrans);
gst_object_unref (audiofilter);
return ret;
}
GST_LOG_OBJECT (filter, "caps: %" GST_PTR_FORMAT, incaps);
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;
if (!gst_ring_buffer_parse_caps (&filter->format, incaps)) {
GST_WARNING_OBJECT (filter, "couldn't parse %" GST_PTR_FORMAT, incaps);
return FALSE;
}
}
static void
gst_audio_filter_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAudioFilter *src;
klass = GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (filter));
g_return_if_fail (GST_IS_AUDIO_FILTER (object));
src = GST_AUDIO_FILTER (object);
if (klass->setup)
ret = klass->setup (filter, &filter->format);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return ret;
}
/**
* gst_audio_filter_class_add_pad_templates:
* @klass: an #GstAudioFilterClass
* @allowed_caps: what formats the filter can handle, as #GstCaps
*
* Convenience function to add pad templates to this element class, with
* @allowed_caps as the caps that can be handled.
*
* This function is usually used from within a GObject base_init function.
*
* Since: 0.10.12
*/
void
gst_audio_filter_class_add_pad_templates (GstAudioFilterClass *
audio_filter_class, const GstCaps * caps)
gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,
const GstCaps * allowed_caps)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (audio_filter_class);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
audio_filter_class->caps = gst_caps_copy (caps);
g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));
g_return_if_fail (allowed_caps != NULL);
g_return_if_fail (GST_IS_CAPS (allowed_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_caps_copy (allowed_caps)));
gst_element_class_add_pad_template (element_class,
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
gst_caps_copy (caps)));
gst_caps_copy (allowed_caps)));
}
......@@ -21,23 +21,15 @@
#ifndef __GST_AUDIO_FILTER_H__
#define __GST_AUDIO_FILTER_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/gstringbuffer.h>
G_BEGIN_DECLS
typedef struct _GstAudioFilter GstAudioFilter;
typedef struct _GstAudioFilterClass GstAudioFilterClass;
typedef void (*GstAudioFilterFilterFunc)(GstAudioFilter *filter,
GstBuffer *outbuf, GstBuffer *inbuf);
typedef void (*GstAudioFilterInplaceFilterFunc)(GstAudioFilter *filter,
GstBuffer *buffer);
typedef void (*GstAudioFilterSetupFunc) (GstAudioFilter *filter);
#define GST_TYPE_AUDIO_FILTER \
(gst_audio_filter_get_type())
#define GST_AUDIO_FILTER(obj) \
......@@ -49,43 +41,48 @@ typedef void (*GstAudioFilterSetupFunc) (GstAudioFilter *filter);
#define GST_IS_AUDIO_FILTER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FILTER))
/**
* GstAudioFilter:
*
* Base class for audio filters with the same format for input and output.
*
* Since: 0.10.12
*/
struct _GstAudioFilter {
GstElement element;
GstBaseTransform basetransform;
GstPad *sinkpad,*srcpad;
/* audio state */
gboolean inited;
gboolean passthru;
int rate;
int width;
int channels;
int depth;
int n_samples;
int size;
int bytes_per_sample;
/*< protected >*/
GstRingBufferSpec format; /* currently configured format */
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstAudioFilterClass:
* @setup: virtual function called whenever the format changes
*
* In addition to the @setup virtual function, you should also override the
* GstBaseTransform::transform and/or GstBaseTransform::transform_ip virtual
* function.
*
* Since: 0.10.12
*/
struct _GstAudioFilterClass {
GstElementClass parent_class;
GstBaseTransformClass basetransformclass;
GstCaps *caps;
GstAudioFilterSetupFunc setup;
GstAudioFilterInplaceFilterFunc filter_inplace;
GstAudioFilterFilterFunc filter;
/* virtual function, called whenever the format changes */
gboolean (*setup) (GstAudioFilter * filter, GstRingBufferSpec * format);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GType gst_audio_filter_get_type(void);
GType gst_audio_filter_get_type (void);
void gst_audio_filter_class_add_pad_templates (GstAudioFilterClass *audiofilterclass, const GstCaps *caps);
void gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,
const GstCaps * caps);
G_END_DECLS
......
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