gstreamer/gst/sine/gstsinesrc.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
* 2001 Steve Baker <stevebaker_org@yahoo.co.uk>
*
* gstsinesrc.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <gst/controller/gstcontroller.h>
#include "gstsinesrc.h"
GstElementDetails gst_sinesrc_details = {
"Sine-wave src",
"Source/Audio",
"Create a sine wave of a given frequency and volume",
"Erik Walthinsen <omega@cse.ogi.edu>"
};
enum
{
PROP_0,
PROP_SAMPLES_PER_BUFFER,
PROP_FREQ,
PROP_VOLUME,
PROP_IS_LIVE,
PROP_TIMESTAMP_OFFSET,
};
static GstStaticPadTemplate gst_sinesrc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, " "rate = (int) [ 1, MAX ], " "channels = (int) 1")
);
GST_BOILERPLATE (GstSineSrc, gst_sinesrc, GstBaseSrc, GST_TYPE_BASE_SRC);
static void gst_sinesrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_sinesrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_sinesrc_unlock (GstBaseSrc * bsrc);
static gboolean gst_sinesrc_setcaps (GstBaseSrc * basesrc, GstCaps * caps);
static void gst_sinesrc_src_fixate (GstPad * pad, GstCaps * caps);
static const GstQueryType *gst_sinesrc_get_query_types (GstPad * pad);
static gboolean gst_sinesrc_src_query (GstPad * pad, GstQuery * query);
static GstFlowReturn gst_sinesrc_create (GstBaseSrc * basesrc, guint64 offset,
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
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guint length, GstBuffer ** buffer);
static gboolean gst_sinesrc_start (GstBaseSrc * basesrc);
static void
gst_sinesrc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_sinesrc_src_template));
gst_element_class_set_details (element_class, &gst_sinesrc_details);
}
static void
gst_sinesrc_class_init (GstSineSrcClass * klass)
{
GObjectClass *gobject_class;
GstBaseSrcClass *gstbasesrc_class;
gobject_class = (GObjectClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
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gobject_class->set_property = gst_sinesrc_set_property;
gobject_class->get_property = gst_sinesrc_get_property;
g_object_class_install_property (gobject_class,
PROP_SAMPLES_PER_BUFFER,
g_param_spec_int ("samplesperbuffer", "Samples per buffer",
"Number of samples in each outgoing buffer",
1, G_MAXINT, 1024, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_FREQ,
g_param_spec_double ("freq", "Frequency", "Frequency of sine source",
0.0, 20000.0, 440.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume", "Volume",
0.0, 1.0, 0.8, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_IS_LIVE,
g_param_spec_boolean ("is-live", "Is Live",
"Whether to act as a live source", FALSE, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_TIMESTAMP_OFFSET,
g_param_spec_int64 ("timestamp-offset", "Timestamp offset",
"An offset added to timestamps set on buffers (in ns)", G_MININT64,
G_MAXINT64, 0, G_PARAM_READWRITE));
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_sinesrc_setcaps);
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_sinesrc_start);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_sinesrc_create);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_sinesrc_unlock);
}
static void
gst_sinesrc_init (GstSineSrc * src, GstSineSrcClass * g_class)
{
GstPad *pad = GST_BASE_SRC_PAD (src);
gst_pad_set_fixatecaps_function (pad, gst_sinesrc_src_fixate);
gst_pad_set_query_function (pad, gst_sinesrc_src_query);
gst_pad_set_query_type_function (pad, gst_sinesrc_get_query_types);
src->samplerate = 44100;
src->volume = 1.0;
src->freq = 440.0;
gst_base_src_set_live (GST_BASE_SRC (src), FALSE);
src->samples_per_buffer = 1024;
src->timestamp = G_GINT64_CONSTANT (0);
src->offset = G_GINT64_CONSTANT (0);
src->timestamp_offset = G_GINT64_CONSTANT (0);
}
static void
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
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gst_sinesrc_src_fixate (GstPad * pad, GstCaps * caps)
{
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
gst_caps_structure_fixate_field_nearest_int (structure, "rate", 44100);
}
static gboolean
gst_sinesrc_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
{
GstSineSrc *sinesrc;
const GstStructure *structure;
gboolean ret;
sinesrc = GST_SINESRC (basesrc);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &sinesrc->samplerate);
return ret;
}
static const GstQueryType *
gst_sinesrc_get_query_types (GstPad * pad)
{
static const GstQueryType query_types[] = {
GST_QUERY_POSITION,
0,
};
return query_types;
}
static gboolean
gst_sinesrc_src_query (GstPad * pad, GstQuery * query)
{
gboolean res = FALSE;
GstSineSrc *src;
src = GST_SINESRC (GST_PAD_PARENT (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gint64 current;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
current = src->timestamp;
res = TRUE;
break;
case GST_FORMAT_DEFAULT: /* samples */
current = src->offset / 2; /* 16bpp audio */
res = TRUE;
break;
case GST_FORMAT_BYTES:
current = src->offset;
res = TRUE;
break;
default:
break;
}
if (res) {
gst_query_set_position (query, format, current);
}
break;
}
case GST_QUERY_DURATION:
{
GstFormat format;
gst_query_parse_duration (query, &format, NULL);
gst_query_set_duration (query, format, -1);
break;
}
default:
break;
}
return res;
}
/* with STREAM_LOCK */
static GstClockReturn
gst_sinesrc_wait (GstSineSrc * src, GstClockTime time)
{
GstClockReturn ret;
GstClockTime base_time;
GST_LOCK (src);
/* clock_id should be NULL outside of this function */
g_assert (src->clock_id == NULL);
g_assert (GST_CLOCK_TIME_IS_VALID (time));
base_time = GST_ELEMENT (src)->base_time;
src->clock_id = gst_clock_new_single_shot_id (GST_ELEMENT_CLOCK (src),
time + base_time);
GST_UNLOCK (src);
ret = gst_clock_id_wait (src->clock_id, NULL);
GST_LOCK (src);
gst_clock_id_unref (src->clock_id);
src->clock_id = NULL;
GST_UNLOCK (src);
return ret;
}
static gboolean
gst_sinesrc_unlock (GstBaseSrc * bsrc)
{
GstSineSrc *src = GST_SINESRC (bsrc);
GST_LOCK (src);
if (src->clock_id)
gst_clock_id_unschedule (src->clock_id);
GST_UNLOCK (src);
return TRUE;
}
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
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static GstFlowReturn
gst_sinesrc_create (GstBaseSrc * basesrc, guint64 offset,
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
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guint length, GstBuffer ** buffer)
{
GstSineSrc *src;
GstBuffer *buf;
guint tdiff;
gdouble step;
gint16 *samples;
gint i;
src = GST_SINESRC (basesrc);
if (!src->tags_pushed) {
GstTagList *taglist;
GstEvent *event;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_DESCRIPTION, "sine wave", NULL);
event = gst_event_new_tag (taglist);
gst_pad_push_event (basesrc->srcpad, event);
src->tags_pushed = TRUE;
}
tdiff = src->samples_per_buffer * GST_SECOND / src->samplerate;
if (gst_base_src_is_live (basesrc)) {
GstClockReturn ret;
ret = gst_sinesrc_wait (src, src->timestamp + src->timestamp_offset);
if (ret == GST_CLOCK_UNSCHEDULED)
goto unscheduled;
}
buf = gst_buffer_new_and_alloc (src->samples_per_buffer * sizeof (gint16));
gst_buffer_set_caps (buf, GST_PAD_CAPS (basesrc->srcpad));
GST_BUFFER_TIMESTAMP (buf) = src->timestamp + src->timestamp_offset;
/* offset is the number of samples */
GST_BUFFER_OFFSET (buf) = src->offset;
GST_BUFFER_OFFSET_END (buf) = src->offset + src->samples_per_buffer;
GST_BUFFER_DURATION (buf) = tdiff;
gst_object_sync_values (G_OBJECT (src), src->timestamp);
samples = (gint16 *) GST_BUFFER_DATA (buf);
src->timestamp += tdiff;
src->offset += src->samples_per_buffer;
step = 2 * M_PI * src->freq / src->samplerate;
for (i = 0; i < src->samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
samples[i] = sin (src->accumulator) * src->volume * 32767.0;
}
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
*buffer = buf;
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
return GST_FLOW_OK;
unscheduled:
{
GST_DEBUG_OBJECT (src, "Unscheduled while waiting for clock");
return GST_FLOW_WRONG_STATE; /* is this the right return? */
}
}
static void
gst_sinesrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSineSrc *src = GST_SINESRC (object);
switch (prop_id) {
case PROP_SAMPLES_PER_BUFFER:
src->samples_per_buffer = g_value_get_int (value);
break;
case PROP_FREQ:
src->freq = g_value_get_double (value);
break;
case PROP_VOLUME:
src->volume = g_value_get_double (value);
break;
case PROP_IS_LIVE:
gst_base_src_set_live (GST_BASE_SRC (src), g_value_get_boolean (value));
break;
case PROP_TIMESTAMP_OFFSET:
src->timestamp_offset = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_sinesrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSineSrc *src = GST_SINESRC (object);
switch (prop_id) {
case PROP_SAMPLES_PER_BUFFER:
g_value_set_int (value, src->samples_per_buffer);
break;
case PROP_FREQ:
g_value_set_double (value, src->freq);
break;
case PROP_VOLUME:
g_value_set_double (value, src->volume);
break;
case PROP_IS_LIVE:
g_value_set_boolean (value, gst_base_src_is_live (GST_BASE_SRC (src)));
break;
case PROP_TIMESTAMP_OFFSET:
g_value_set_int64 (value, src->timestamp_offset);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_sinesrc_start (GstBaseSrc * basesrc)
{
GstSineSrc *src = GST_SINESRC (basesrc);
src->timestamp = G_GINT64_CONSTANT (0);
src->offset = G_GINT64_CONSTANT (0);
return TRUE;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "sinesrc",
GST_RANK_NONE, GST_TYPE_SINESRC);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"sine",
"Sine audio wave generator",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)