borgify further clean up docs a little

Original commit message from CVS:
borgify further
clean up docs a little
parent 109cd71c
2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-libvisual.xml:
* gst/audioconvert/plugin.h:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_get_type),
(gst_audio_rate_base_init), (gst_audio_rate_class_init),
(gst_audio_rate_setcaps), (gst_audio_rate_init),
(gst_audio_rate_chain), (gst_audio_rate_set_property),
(gst_audio_rate_get_property), (gst_audio_rate_change_state),
(plugin_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_base_init),
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_get_query_types),
(gst_audio_test_src_src_query), (gst_audio_test_src_create_sine),
(gst_audio_test_src_create_square),
(gst_audio_test_src_create_saw),
(gst_audio_test_src_create_triangle),
(gst_audio_test_src_create_silence),
(gst_audio_test_src_create_white_noise),
(gst_audio_test_src_init_pink_noise),
(gst_audio_test_src_generate_pink_noise_value),
(gst_audio_test_src_create_pink_noise),
(gst_audio_test_src_change_wave), (gst_audio_test_src_get_times),
(gst_audio_test_src_create), (gst_audio_test_src_set_property),
(gst_audio_test_src_get_property), (gst_audio_test_src_start),
(plugin_init):
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/subparse/gstsubparse.c: (gst_sub_parse_get_type),
(gst_sub_parse_base_init), (gst_sub_parse_class_init),
(gst_sub_parse_init), (gst_sub_parse_formats),
(gst_sub_parse_src_eventmask), (gst_sub_parse_src_event),
(convert_encoding), (get_next_line),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (feed_textbuf), (handle_buffer),
(gst_sub_parse_loop), (gst_sub_parse_chain),
(gst_sub_parse_change_state), (gst_sub_parse_type_find),
(plugin_init):
* gst/subparse/gstsubparse.h:
* gst/videorate/gstvideorate.c: (gst_video_rate_get_type),
(gst_video_rate_base_init), (gst_video_rate_class_init),
(gst_video_rate_transformcaps), (gst_video_rate_getcaps),
(gst_video_rate_setcaps), (gst_video_rate_blank_data),
(gst_video_rate_init), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_set_property),
(gst_video_rate_get_property), (gst_video_rate_change_state),
(plugin_init):
* gst/videoscale/gstvideoscale.c:
(gst_video_scale_method_get_type), (gst_video_scale_get_capslist),
(gst_video_scale_src_template_factory),
(gst_video_scale_sink_template_factory),
(gst_video_scale_get_type), (gst_video_scale_base_init),
(gst_video_scale_class_init), (gst_video_scale_init),
(gst_video_scale_set_property), (gst_video_scale_get_property),
(gst_video_scale_transform_caps), (gst_video_scale_get_format),
(gst_video_scale_prepare_size), (parse_caps),
(gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
(gst_video_scale_fixate_caps), (gst_video_scale_prepare_image),
(gst_video_scale_transform), (gst_video_scale_handle_src_event),
(plugin_init):
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_pattern_get_type),
(gst_video_test_src_base_init), (gst_video_test_src_class_init),
(gst_video_test_src_init), (gst_video_test_src_src_fixate),
(gst_video_test_src_set_pattern),
(gst_video_test_src_set_property),
(gst_video_test_src_get_property), (gst_video_test_src_getcaps),
(gst_video_test_src_parse_caps), (gst_video_test_src_setcaps),
(gst_video_test_src_event), (gst_video_test_src_get_times),
(gst_video_test_src_create), (plugin_init):
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_get_size),
(gst_video_test_src_smpte), (gst_video_test_src_snow),
(gst_video_test_src_black):
* gst/videotestsrc/videotestsrc.h:
borgify further
clean up docs a little
2005-11-30 Wim Taymans <wim@fluendo.com> 2005-11-30 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/rtp/gstbasertpdepayload.h: * gst-libs/gst/rtp/gstbasertpdepayload.h:
......
...@@ -14,6 +14,9 @@ a support library for audio elements ...@@ -14,6 +14,9 @@ a support library for audio elements
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
<!-- ##### ENUM GstAudioFieldFlag ##### --> <!-- ##### ENUM GstAudioFieldFlag ##### -->
<para> <para>
......
...@@ -14,6 +14,9 @@ interface for adjusting color balance settings ...@@ -14,6 +14,9 @@ interface for adjusting color balance settings
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
<!-- ##### STRUCT GstColorBalance ##### --> <!-- ##### STRUCT GstColorBalance ##### -->
<para> <para>
......
...@@ -14,3 +14,6 @@ gconf default elements support ...@@ -14,3 +14,6 @@ gconf default elements support
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
...@@ -14,6 +14,9 @@ interface for elements that provide mixer operations ...@@ -14,6 +14,9 @@ interface for elements that provide mixer operations
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
<!-- ##### STRUCT GstMixer ##### --> <!-- ##### STRUCT GstMixer ##### -->
<para> <para>
......
...@@ -14,6 +14,9 @@ an implementation of an audio ringbuffer ...@@ -14,6 +14,9 @@ an implementation of an audio ringbuffer
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
<!-- ##### STRUCT GstRingBuffer ##### --> <!-- ##### STRUCT GstRingBuffer ##### -->
<para> <para>
......
...@@ -14,6 +14,9 @@ interface for elements that provide tuner operations ...@@ -14,6 +14,9 @@ interface for elements that provide tuner operations
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
<!-- ##### STRUCT GstTuner ##### --> <!-- ##### STRUCT GstTuner ##### -->
<para> <para>
......
...@@ -14,6 +14,9 @@ interface for setting/getting a Window on elements supporting it. ...@@ -14,6 +14,9 @@ interface for setting/getting a Window on elements supporting it.
</para> </para>
<!-- ##### SECTION Stability_Level ##### -->
<!-- ##### STRUCT GstXOverlay ##### --> <!-- ##### STRUCT GstXOverlay ##### -->
<para> <para>
......
...@@ -80,6 +80,7 @@ EXTRA_HFILES = \ ...@@ -80,6 +80,7 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/theora/gsttheoraenc.h \ $(top_srcdir)/ext/theora/gsttheoraenc.h \
$(top_srcdir)/ext/vorbis/vorbisenc.h \ $(top_srcdir)/ext/vorbis/vorbisenc.h \
$(top_srcdir)/gst/audioconvert/gstaudioconvert.h \ $(top_srcdir)/gst/audioconvert/gstaudioconvert.h \
$(top_srcdir)/gst/audiotestsrc/gstaudiotestsrc.h \
$(top_srcdir)/gst/ffmpegcolorspace/gstffmpegcolorspace.h \ $(top_srcdir)/gst/ffmpegcolorspace/gstffmpegcolorspace.h \
$(top_srcdir)/gst/tcp/gstmultifdsink.h \ $(top_srcdir)/gst/tcp/gstmultifdsink.h \
$(top_srcdir)/gst/tcp/gsttcpserversink.h \ $(top_srcdir)/gst/tcp/gsttcpserversink.h \
......
...@@ -1008,3 +1008,123 @@ ...@@ -1008,3 +1008,123 @@
<DEFAULT>Sine</DEFAULT> <DEFAULT>Sine</DEFAULT>
</ARG> </ARG>
<ARG>
<NAME>GstVideoScale::method</NAME>
<TYPE>GstVideoScaleMethod</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>method</NICK>
<BLURB>method.</BLURB>
<DEFAULT>Nearest Neighbour</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoRate::drop</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Drop</NICK>
<BLURB>Number of dropped frames.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoRate::duplicate</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Duplicate</NICK>
<BLURB>Number of duplicated frames.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoRate::in</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>In</NICK>
<BLURB>Number of input frames.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoRate::new-pref</NAME>
<TYPE>gdouble</TYPE>
<RANGE>[0,1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>New Pref</NICK>
<BLURB>Value indicating how much to prefer new frames.</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoRate::out</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Out</NICK>
<BLURB>Number of output frames.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoRate::silent</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>silent</NICK>
<BLURB>Don't emit notify for dropped and duplicated frames.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioRate::add</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Add</NICK>
<BLURB>Number of added samples.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioRate::drop</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Drop</NICK>
<BLURB>Number of dropped samples.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioRate::in</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>In</NICK>
<BLURB>Number of input samples.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioRate::out</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Out</NICK>
<BLURB>Number of output samples.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioRate::silent</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>silent</NICK>
<BLURB>Don't emit notify for dropped and duplicated frames.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
...@@ -3,10 +3,10 @@ ...@@ -3,10 +3,10 @@
<description>libvisual visualization plugins</description> <description>libvisual visualization plugins</description>
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename> <filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
<basename>libgstlibvisual.so</basename> <basename>libgstlibvisual.so</basename>
<version>0.9.6</version> <version>0.9.6.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>
......
...@@ -20,8 +20,8 @@ ...@@ -20,8 +20,8 @@
*/ */
#ifndef __GST_PLUGIN_AUDIOCONVERT_H__ #ifndef __GST_PLUGIN_AUDIO_CONVERT_H__
#define __GST_PLUGIN_AUDIOCONVERT_H__ #define __GST_PLUGIN_AUDIO_CONVERT_H__
#include <gst/gst.h> #include <gst/gst.h>
...@@ -32,4 +32,4 @@ GType gst_audio_convert_get_type (void); ...@@ -32,4 +32,4 @@ GType gst_audio_convert_get_type (void);
G_END_DECLS G_END_DECLS
#endif /* __GST_PLUGIN_AUDIOCONVERT_H__ */ #endif /* __GST_PLUGIN_AUDIO_CONVERT_H__ */
...@@ -25,21 +25,21 @@ ...@@ -25,21 +25,21 @@
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/audio.h> #include <gst/audio/audio.h>
#define GST_TYPE_AUDIORATE \ #define GST_TYPE_AUDIO_RATE \
(gst_audiorate_get_type()) (gst_audio_rate_get_type())
#define GST_AUDIORATE(obj) \ #define GST_AUDIO_RATE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORATE,GstAudiorate)) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RATE,GstAudioRate))
#define GST_AUDIORATE_CLASS(klass) \ #define GST_AUDIO_RATE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORATE,GstAudiorate)) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RATE,GstAudioRate))
#define GST_IS_AUDIORATE(obj) \ #define GST_IS_AUDIO_RATE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORATE)) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RATE))
#define GST_IS_AUDIORATE_CLASS(obj) \ #define GST_IS_AUDIO_RATE_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORATE)) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RATE))
typedef struct _GstAudiorate GstAudiorate; typedef struct _GstAudioRate GstAudioRate;
typedef struct _GstAudiorateClass GstAudiorateClass; typedef struct _GstAudioRateClass GstAudioRateClass;
struct _GstAudiorate struct _GstAudioRate
{ {
GstElement element; GstElement element;
...@@ -54,19 +54,19 @@ struct _GstAudiorate ...@@ -54,19 +54,19 @@ struct _GstAudiorate
gboolean silent; gboolean silent;
}; };
struct _GstAudiorateClass struct _GstAudioRateClass
{ {
GstElementClass parent_class; GstElementClass parent_class;
}; };
/* elementfactory information */ /* elementfactory information */
static GstElementDetails audiorate_details = static GstElementDetails audio_rate_details =
GST_ELEMENT_DETAILS ("Audio rate adjuster", GST_ELEMENT_DETAILS ("Audio rate adjuster",
"Filter/Effect/Audio", "Filter/Effect/Audio",
"Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream", "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
"Wim Taymans <wim@fluendo.com>"); "Wim Taymans <wim@fluendo.com>");
/* GstAudiorate signals and args */ /* GstAudioRate signals and args */
enum enum
{ {
/* FILL ME */ /* FILL ME */
...@@ -86,84 +86,84 @@ enum ...@@ -86,84 +86,84 @@ enum
/* FILL ME */ /* FILL ME */
}; };
static GstStaticPadTemplate gst_audiorate_src_template = static GstStaticPadTemplate gst_audio_rate_src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_SRC,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS) GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
); );
static GstStaticPadTemplate gst_audiorate_sink_template = static GstStaticPadTemplate gst_audio_rate_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS) GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
); );
static void gst_audiorate_base_init (gpointer g_class); static void gst_audio_rate_base_init (gpointer g_class);
static void gst_audiorate_class_init (GstAudiorateClass * klass); static void gst_audio_rate_class_init (GstAudioRateClass * klass);
static void gst_audiorate_init (GstAudiorate * audiorate); static void gst_audio_rate_init (GstAudioRate * audiorate);
static GstFlowReturn gst_audiorate_chain (GstPad * pad, GstBuffer * buf); static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);
static void gst_audiorate_set_property (GObject * object, static void gst_audio_rate_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec); guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audiorate_get_property (GObject * object, static void gst_audio_rate_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec); guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_audiorate_change_state (GstElement * element, static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element,
GstStateChange transition); GstStateChange transition);
static GstElementClass *parent_class = NULL; static GstElementClass *parent_class = NULL;
/*static guint gst_audiorate_signals[LAST_SIGNAL] = { 0 }; */ /*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
static GType static GType
gst_audiorate_get_type (void) gst_audio_rate_get_type (void)
{ {
static GType audiorate_type = 0; static GType audio_rate_type = 0;
if (!audiorate_type) { if (!audio_rate_type) {
static const GTypeInfo audiorate_info = { static const GTypeInfo audio_rate_info = {
sizeof (GstAudiorateClass), sizeof (GstAudioRateClass),
gst_audiorate_base_init, gst_audio_rate_base_init,
NULL, NULL,
(GClassInitFunc) gst_audiorate_class_init, (GClassInitFunc) gst_audio_rate_class_init,
NULL, NULL,
NULL, NULL,
sizeof (GstAudiorate), sizeof (GstAudioRate),
0, 0,
(GInstanceInitFunc) gst_audiorate_init, (GInstanceInitFunc) gst_audio_rate_init,
}; };
audiorate_type = g_type_register_static (GST_TYPE_ELEMENT, audio_rate_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstAudiorate", &audiorate_info, 0); "GstAudioRate", &audio_rate_info, 0);
} }
return audiorate_type; return audio_rate_type;
} }
static void static void
gst_audiorate_base_init (gpointer g_class) gst_audio_rate_base_init (gpointer g_class)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &audiorate_details); gst_element_class_set_details (element_class, &audio_rate_details);
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audiorate_sink_template)); gst_static_pad_template_get (&gst_audio_rate_sink_template));
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audiorate_src_template)); gst_static_pad_template_get (&gst_audio_rate_src_template));
} }
static void static void
gst_audiorate_class_init (GstAudiorateClass * klass) gst_audio_rate_class_init (GstAudioRateClass * klass)
{ {
GObjectClass *object_class = G_OBJECT_CLASS (klass); GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass); parent_class = g_type_class_peek_parent (klass);
object_class->set_property = gst_audiorate_set_property; object_class->set_property = gst_audio_rate_set_property;
object_class->get_property = gst_audiorate_get_property; object_class->get_property = gst_audio_rate_get_property;
g_object_class_install_property (object_class, ARG_IN, g_object_class_install_property (object_class, ARG_IN,
g_param_spec_uint64 ("in", "In", g_param_spec_uint64 ("in", "In",
...@@ -182,18 +182,18 @@ gst_audiorate_class_init (GstAudiorateClass * klass) ...@@ -182,18 +182,18 @@ gst_audiorate_class_init (GstAudiorateClass * klass)
"Don't emit notify for dropped and duplicated frames", "Don't emit notify for dropped and duplicated frames",
DEFAULT_SILENT, G_PARAM_READWRITE)); DEFAULT_SILENT, G_PARAM_READWRITE));
element_class->change_state = gst_audiorate_change_state; element_class->change_state = gst_audio_rate_change_state;
} }
static gboolean static gboolean
gst_audiorate_setcaps (GstPad * pad, GstCaps * caps) gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)
{ {
GstAudiorate *audiorate; GstAudioRate *audiorate;
GstStructure *structure; GstStructure *structure;
GstPad *otherpad; GstPad *otherpad;
gint ret, channels, depth; gint ret, channels, depth;
audiorate = GST_AUDIORATE (gst_pad_get_parent (pad)); audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad : otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
audiorate->srcpad; audiorate->srcpad;
...@@ -217,19 +217,19 @@ gst_audiorate_setcaps (GstPad * pad, GstCaps * caps) ...@@ -217,19 +217,19 @@ gst_audiorate_setcaps (GstPad * pad, GstCaps * caps)
} }
static void static void
gst_audiorate_init (GstAudiorate * audiorate) gst_audio_rate_init (GstAudioRate * audiorate)
{ {
audiorate->sinkpad = audiorate->sinkpad =
gst_pad_new_from_static_template (&gst_audiorate_sink_template, "sink"); gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink");
gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad); gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
gst_pad_set_chain_function (audiorate->sinkpad, gst_audiorate_chain); gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain);
gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audiorate_setcaps); gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_setcaps);
gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps); gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
audiorate->srcpad = audiorate->srcpad =
gst_pad_new_from_static_template (&gst_audiorate_src_template, "src"); gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src");
gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad); gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
gst_pad_set_setcaps_function (audiorate->srcpad, gst_audiorate_setcaps); gst_pad_set_setcaps_function (audiorate->srcpad, gst_audio_rate_setcaps);
gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps); gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
audiorate->bytes_per_sample = 1; audiorate->bytes_per_sample = 1;
...@@ -241,15 +241,15 @@ gst_audiorate_init (GstAudiorate * audiorate) ...@@ -241,15 +241,15 @@ gst_audiorate_init (GstAudiorate * audiorate)
} }
static GstFlowReturn static GstFlowReturn
gst_audiorate_chain (GstPad * pad, GstBuffer * buf) gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{ {
GstAudiorate *audiorate; GstAudioRate *audiorate;
GstClockTime in_time, in_duration; GstClockTime in_time, in_duration;
guint64 in_offset, in_offset_end; guint64 in_offset, in_offset_end;
gint in_size; gint in_size;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
audiorate = GST_AUDIORATE (gst_pad_get_parent (pad)); audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));