mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-08-04 04:35:06 +00:00
Code Issues 1 Projects Releases Wiki Activity

ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed.

Browse source 
Original commit message from CVS:
* ext/jack/gstjackaudiosrc.c:
Try committing this once again. Now properly renamed.
This commit is contained in:
Stefan Kost 2008-08-07 13:15:21 +00:00
parent f1d5e92f33
commit 2e1e65e154
2 changed files with 22 additions and 17 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

5
ChangeLog
View file

@ -1,3 +1,8 @@
2008-08-07 Stefan Kost <ensonic@users.sf.net>
* ext/jack/gstjackaudiosrc.c:
Try committing this once again. Now properly renamed.
2008-08-07 Stefan Kost <ensonic@users.sf.net> 2008-08-07 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/Makefile.am: * docs/plugins/Makefile.am:

34
ext/jack/gstjackaudiosrc.c
View file

@ -41,7 +41,7 @@
*/ */
/** /**
* SECTION:element-jackaudiosrc * SECTION:element-jack_audio_src
* @see_also: #GstBaseAudioSrc, #GstRingBuffer * @see_also: #GstBaseAudioSrc, #GstRingBuffer
* *
* A Src that inputs data from Jack ports. * A Src that inputs data from Jack ports.
@ -82,7 +82,7 @@
#include "gstjackringbuffer.h" #include "gstjackringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug); GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
#define GST_CAT_DEFAULT gst_jackaudiosrc_debug #define GST_CAT_DEFAULT gst_jack_audio_src_debug
static gboolean static gboolean
gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels) gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
@ -646,24 +646,24 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
#define _do_init(bla) \ #define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element"); GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc, GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init); GST_TYPE_BASE_AUDIO_SRC, _do_init);
static void gst_jackaudiosrc_set_property (GObject * object, guint prop_id, static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec); const GValue * value, GParamSpec * pspec);
static void gst_jackaudiosrc_get_property (GObject * object, guint prop_id, static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec); GValue * value, GParamSpec * pspec);
static GstCaps *gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc); static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc);
static GstRingBuffer *gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
src); src);
/* GObject vmethod implementations */ /* GObject vmethod implementations */
static void static void
gst_jackaudiosrc_base_init (gpointer gclass) gst_jack_audio_src_base_init (gpointer gclass)
{ {
static GstElementDetails gst_jackaudiosrc_details = { static GstElementDetails gst_jack_audio_src_details = {
"Audio Source (Jack)", "Audio Source (Jack)",
"Source/Audio", "Source/Audio",
"Input from Jack", "Input from Jack",
@ -673,12 +673,12 @@ gst_jackaudiosrc_base_init (gpointer gclass)
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory)); gst_static_pad_template_get (&src_factory));
gst_element_class_set_details (element_class, &gst_jackaudiosrc_details); gst_element_class_set_details (element_class, &gst_jack_audio_src_details);
} }
/* initialize the jackaudiosrc's class */ /* initialize the jack_audio_src's class */
static void static void
gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass) gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class; GstElementClass *gstelement_class;
@ -723,7 +723,7 @@ gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass)
* initialize instance structure * initialize instance structure
*/ */
static void static void
gst_jackaudiosrc_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
{ {
//gst_base_src_set_live(GST_BASE_SRC (src), TRUE); //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
src->connect = DEFAULT_PROP_CONNECT; src->connect = DEFAULT_PROP_CONNECT;
@ -733,7 +733,7 @@ gst_jackaudiosrc_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
} }
static void static void
gst_jackaudiosrc_set_property (GObject * object, guint prop_id, gst_jack_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec) const GValue * value, GParamSpec * pspec)
{ {
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
@ -753,7 +753,7 @@ gst_jackaudiosrc_set_property (GObject * object, guint prop_id,
} }
static void static void
gst_jackaudiosrc_get_property (GObject * object, guint prop_id, gst_jack_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec) GValue * value, GParamSpec * pspec)
{ {
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
@ -772,7 +772,7 @@ gst_jackaudiosrc_get_property (GObject * object, guint prop_id,
} }
static GstCaps * static GstCaps *
gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc) gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
{ {
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc); GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
const char **ports; const char **ports;
@ -829,7 +829,7 @@ no_client:
} }
static GstRingBuffer * static GstRingBuffer *
gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * src) gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{ {
GstRingBuffer *buffer; GstRingBuffer *buffer;