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add new plugin and element

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Original commit message from CVS:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type):
* gst/audiotestsrc/Makefile.am:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_base_init),
(gst_audiotestsrc_class_init), (gst_audiotestsrc_init),
(gst_audiotestsrc_src_fixate), (gst_audiotestsrc_setcaps),
(gst_audiotestsrc_get_query_types), (gst_audiotestsrc_src_query),
(gst_audiotestsrc_wait), (gst_audiotestsrc_unlock),
(gst_audiotestsrc_create_sine), (gst_audiotestsrc_create_square),
(gst_audiotestsrc_create_saw), (gst_audiotestsrc_create_triangle),
(gst_audiotestsrc_create_silence),
(gst_audiotestsrc_create_white_noise),
(gst_audiotestsrc_change_wave), (gst_audiotestsrc_create),
(gst_audiotestsrc_set_property), (gst_audiotestsrc_get_property),
(gst_audiotestsrc_start), (plugin_init):
* gst/audiotestsrc/gstaudiotestsrc.h:
add new plugin and element
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init):
use gobject_class
This commit is contained in:
Stefan Kost 2005-10-09 18:34:44 +00:00
parent 23375d3a0e
commit 9be025e197
9 changed files with 746 additions and 16 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

25
ChangeLog
View file

@ -1,3 +1,28 @@
2005-10-09 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type):
* gst/audiotestsrc/Makefile.am:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_base_init),
(gst_audiotestsrc_class_init), (gst_audiotestsrc_init),
(gst_audiotestsrc_src_fixate), (gst_audiotestsrc_setcaps),
(gst_audiotestsrc_get_query_types), (gst_audiotestsrc_src_query),
(gst_audiotestsrc_wait), (gst_audiotestsrc_unlock),
(gst_audiotestsrc_create_sine), (gst_audiotestsrc_create_square),
(gst_audiotestsrc_create_saw), (gst_audiotestsrc_create_triangle),
(gst_audiotestsrc_create_silence),
(gst_audiotestsrc_create_white_noise),
(gst_audiotestsrc_change_wave), (gst_audiotestsrc_create),
(gst_audiotestsrc_set_property), (gst_audiotestsrc_get_property),
(gst_audiotestsrc_start), (plugin_init):
* gst/audiotestsrc/gstaudiotestsrc.h:
add new plugin and element
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init):
use gobject_class
2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_class_init),

12
configure.ac
View file

@ -293,18 +293,19 @@ dnl these are all the gst plug-ins, compilable without additional libs
GST_PLUGINS_ALL="\
adder \
audioconvert \
audiorate \
audioresample \
audiorate \
audioresample \
audiotestsrc \
ffmpegcolorspace \
playback \
sine \
subparse \
tcp \
subparse \
tcp \
typefind \
videotestsrc \
videorate \
videoscale \
volume \
volume \
"
dnl see if we can build C++ plug-ins
@ -768,6 +769,7 @@ gst/adder/Makefile
gst/audioconvert/Makefile
gst/audiorate/Makefile
gst/audioresample/Makefile
gst/audiotestsrc/Makefile
gst/ffmpegcolorspace/Makefile
gst/playback/Makefile
gst/sine/Makefile

2
docs/plugins/gst-plugins-base-plugins-docs.sgml
View file

@ -13,6 +13,7 @@
<chapter>
<title>gst-plugins-base Elements</title>
<xi:include href="xml/element-audioconvert.xml" />
<xi:include href="xml/element-audiotestsrc.xml" />
<xi:include href="xml/element-ffmpegcolorspace.xml" />
<!--
<xi:include href="xml/element-gnomevfssink.xml" />
@ -31,6 +32,7 @@
<xi:include href="xml/plugin-alsa.xml" />
<xi:include href="xml/plugin-audioconvert.xml" />
<xi:include href="xml/plugin-audiorate.xml" />
<xi:include href="xml/plugin-audiotestsrc.xml" />
<xi:include href="xml/plugin-decodebin.xml" />
<xi:include href="xml/plugin-gnomevfs.xml" />
<xi:include href="xml/plugin-libvisual.xml" />

8
docs/plugins/gst-plugins-base-plugins-sections.txt
View file

@ -7,6 +7,14 @@ GstAudioConvert
GstAudioConvertClass
</SECTION>
<SECTION>
<FILE>element-audiotestsrc</FILE>
<TITLE>audiotestsrc</TITLE>
GstAudioTestSrc
<SUBSECTION Standard>
GstAudioTestSrcClass
</SECTION>
<SECTION>
<FILE>element-ffmpegcolorspace</FILE>
<TITLE>ffmpegcolorspace</TITLE>

14
gst/audioscale/gstaudioscale.c
View file

@ -85,16 +85,10 @@ gst_audioscale_method_get_type (void)
{
static GType audioscale_method_type = 0;
static GEnumValue audioscale_methods[] = {
{
GST_RESAMPLE_NEAREST, "0", "Nearest"}
,
{
GST_RESAMPLE_BILINEAR, "1", "Bilinear"}
, {
GST_RESAMPLE_SINC, "2", "Sinc"}
, {
0, NULL, NULL}
,
{GST_RESAMPLE_NEAREST, "0", "Nearest"},
{GST_RESAMPLE_BILINEAR, "1", "Bilinear"},
{GST_RESAMPLE_SINC, "2", "Sinc"},
{0, NULL, NULL},
};
if (!audioscale_method_type) {

8
gst/audiotestsrc/Makefile.am Normal file
View file

@ -0,0 +1,8 @@
plugin_LTLIBRARIES = libgstaudiotestsrc.la
libgstaudiotestsrc_la_SOURCES = gstaudiotestsrc.c
libgstaudiotestsrc_la_CFLAGS = $(GST_CFLAGS)
libgstaudiotestsrc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudiotestsrc_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(GST_CTRL_LIBS)
noinst_HEADERS = gstaudiotestsrc.h

598
gst/audiotestsrc/gstaudiotestsrc.c Normal file
View file

@ -0,0 +1,598 @@
/* GStreamer
* Copyright (C) 2005 Stefan Kost <ensonic@users.sf.net>
*
* gstaudiotestsrc.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.
*/
/**
* SECTION:element-audiotestsrc
*
* <refsect2>
* AudioTestSrc can be used to generate basic audio signals. It support several
* different waveforms, variable pitch and volume.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc ! audioconvert ! alsasink
* </programlisting>
* This pipeline produces a sine with default pitch and volume.
* </para>
* <para>
* <programlisting>
* gst-launch audiotestsrc wave=2 freq=200 ! audioconvert ! tee name=t ! alsasink t. ! libvisual_lv_scope ! ffmpegcolorspace ! xvimagesink
* </programlisting>
* In this example a saw-wave has been choosen. The wave is shown using a scope.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <gst/controller/gstcontroller.h>
#include "gstaudiotestsrc.h"
GstElementDetails gst_audiotestsrc_details = {
"Audio test source",
"Source/Audio",
"Creates audio test signals of given frequency and volume",
"Stefan Kost <ensonic@users.sf.net>"
};
enum
{
PROP_0,
PROP_SAMPLES_PER_BUFFER,
PROP_WAVE,
PROP_FREQ,
PROP_VOLUME,
PROP_IS_LIVE,
PROP_TIMESTAMP_OFFSET,
};
static GstStaticPadTemplate gst_audiotestsrc_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 (GstAudioTestSrc, gst_audiotestsrc, GstBaseSrc,
GST_TYPE_BASE_SRC);
#define GST_TYPE_AUDIOTESTSRC_WAVE (gst_audiostestsrc_wave_get_type())
static GType
gst_audiostestsrc_wave_get_type (void)
{
static GType audiostestsrc_wave_type = 0;
static GEnumValue audiostestsrc_waves[] = {
{GST_AUDIOTESTSRC_WAVE_SINE, "0", "Sine"},
{GST_AUDIOTESTSRC_WAVE_SQUARE, "1", "Square"},
{GST_AUDIOTESTSRC_WAVE_SAW, "2", "Saw"},
{GST_AUDIOTESTSRC_WAVE_TRIANGLE, "3", "Trinagle"},
{GST_AUDIOTESTSRC_WAVE_SILENCE, "4", "Silence"},
{GST_AUDIOTESTSRC_WAVE_WHITE_NOISE, "5", "White noise"},
{0, NULL, NULL},
};
if (!audiostestsrc_wave_type) {
audiostestsrc_wave_type = g_enum_register_static ("GstAudioTestSrcWave",
audiostestsrc_waves);
}
return audiostestsrc_wave_type;
}
static void gst_audiotestsrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audiotestsrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audiotestsrc_unlock (GstBaseSrc * bsrc);
static gboolean gst_audiotestsrc_setcaps (GstBaseSrc * basesrc, GstCaps * caps);
static void gst_audiotestsrc_src_fixate (GstPad * pad, GstCaps * caps);
static const GstQueryType *gst_audiotestsrc_get_query_types (GstPad * pad);
static gboolean gst_audiotestsrc_src_query (GstPad * pad, GstQuery * query);
static void gst_audiotestsrc_change_wave (GstAudioTestSrc * src);
static GstFlowReturn gst_audiotestsrc_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
static gboolean gst_audiotestsrc_start (GstBaseSrc * basesrc);
static void
gst_audiotestsrc_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_audiotestsrc_src_template));
gst_element_class_set_details (element_class, &gst_audiotestsrc_details);
}
static void
gst_audiotestsrc_class_init (GstAudioTestSrcClass * klass)
{
GObjectClass *gobject_class;
GstBaseSrcClass *gstbasesrc_class;
gobject_class = (GObjectClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gobject_class->set_property = gst_audiotestsrc_set_property;
gobject_class->get_property = gst_audiotestsrc_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_WAVE, g_param_spec_enum ("wave", "Waveform", "Oscillator waveform", GST_TYPE_AUDIOTESTSRC_WAVE, /* enum type */
GST_AUDIOTESTSRC_WAVE_SINE, /* default value */
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_FREQ,
g_param_spec_double ("freq", "Frequency", "Frequency of test signal",
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 of test signal",
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_audiotestsrc_setcaps);
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_audiotestsrc_start);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audiotestsrc_create);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_audiotestsrc_unlock);
}
static void
gst_audiotestsrc_init (GstAudioTestSrc * src, GstAudioTestSrcClass * g_class)
{
GstPad *pad = GST_BASE_SRC_PAD (src);
gst_pad_set_fixatecaps_function (pad, gst_audiotestsrc_src_fixate);
gst_pad_set_query_function (pad, gst_audiotestsrc_src_query);
gst_pad_set_query_type_function (pad, gst_audiotestsrc_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);
src->wave = GST_AUDIOTESTSRC_WAVE_SINE;
gst_audiotestsrc_change_wave (src);
}
static void
gst_audiotestsrc_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_audiotestsrc_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
{
GstAudioTestSrc *audiotestsrc;
const GstStructure *structure;
gboolean ret;
audiotestsrc = GST_AUDIOTESTSRC (basesrc);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &audiotestsrc->samplerate);
return ret;
}
static const GstQueryType *
gst_audiotestsrc_get_query_types (GstPad * pad)
{
static const GstQueryType query_types[] = {
GST_QUERY_POSITION,
0,
};
return query_types;
}
static gboolean
gst_audiotestsrc_src_query (GstPad * pad, GstQuery * query)
{
gboolean res = FALSE;
GstAudioTestSrc *src;
src = GST_AUDIOTESTSRC (GST_PAD_PARENT (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gint64 current;
gst_query_parse_position (query, &format, NULL, 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, -1);
}
break;
}
default:
break;
}
return res;
}
/* with STREAM_LOCK */
static GstClockReturn
gst_audiotestsrc_wait (GstAudioTestSrc * 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_audiotestsrc_unlock (GstBaseSrc * bsrc)
{
GstAudioTestSrc *src = GST_AUDIOTESTSRC (bsrc);
GST_LOCK (src);
if (src->clock_id)
gst_clock_id_unschedule (src->clock_id);
GST_UNLOCK (src);
return TRUE;
}
static void
gst_audiotestsrc_create_sine (GstAudioTestSrc * src, gint16 * samples)
{
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
amp = src->volume * 32767.0;
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] = (gint16) (sin (src->accumulator) * amp);
}
}
static void
gst_audiotestsrc_create_square (GstAudioTestSrc * src, gint16 * samples)
{
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
amp = src->volume * 32767.0;
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] = (gint16) ((src->accumulator < M_PI) ? amp : -amp);
}
}
static void
gst_audiotestsrc_create_saw (GstAudioTestSrc * src, gint16 * samples)
{
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
amp = (src->volume * 32767.0) / M_PI;
for (i = 0; i < src->samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
if (src->accumulator < M_PI) {
samples[i] = (gint16) (src->accumulator * amp);
} else {
samples[i] = (gint16) ((2 * M_PI - src->accumulator) * -amp);
}
}
}
static void
gst_audiotestsrc_create_triangle (GstAudioTestSrc * src, gint16 * samples)
{
gint i;
gdouble step, amp;
step = 2 * M_PI * src->freq / src->samplerate;
amp = (src->volume * 32767.0) / (M_PI * 0.5);
for (i = 0; i < src->samples_per_buffer; i++) {
src->accumulator += step;
if (src->accumulator >= 2 * M_PI)
src->accumulator -= 2 * M_PI;
if (src->accumulator < (M_PI * 0.5)) {
samples[i] = (gint16) (src->accumulator * amp);
} else if (src->accumulator < (M_PI * 1.5)) {
samples[i] = (gint16) ((src->accumulator - M_PI) * -amp);
} else {
samples[i] = (gint16) ((2 * M_PI - src->accumulator) * -amp);
}
}
}
static void
gst_audiotestsrc_create_silence (GstAudioTestSrc * src, gint16 * samples)
{
memset (samples, 0, src->samples_per_buffer * sizeof (gint16));
}
static void
gst_audiotestsrc_create_white_noise (GstAudioTestSrc * src, gint16 * samples)
{
gint i;
gdouble amp;
amp = src->volume * 65535.0;
for (i = 0; i < src->samples_per_buffer; i++) {
samples[i] = (gint16) (32768 - (amp * rand () / (RAND_MAX + 1.0)));
}
}
static void
gst_audiotestsrc_change_wave (GstAudioTestSrc * src)
{
switch (src->wave) {
case GST_AUDIOTESTSRC_WAVE_SINE:
src->process = gst_audiotestsrc_create_sine;
break;
case GST_AUDIOTESTSRC_WAVE_SQUARE:
src->process = gst_audiotestsrc_create_square;
break;
case GST_AUDIOTESTSRC_WAVE_SAW:
src->process = gst_audiotestsrc_create_saw;
break;
case GST_AUDIOTESTSRC_WAVE_TRIANGLE:
src->process = gst_audiotestsrc_create_triangle;
break;
case GST_AUDIOTESTSRC_WAVE_SILENCE:
src->process = gst_audiotestsrc_create_silence;
break;
case GST_AUDIOTESTSRC_WAVE_WHITE_NOISE:
src->process = gst_audiotestsrc_create_white_noise;
break;
default:
GST_ERROR ("invalid wave-form");
break;
}
}
static GstFlowReturn
gst_audiotestsrc_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstAudioTestSrc *src;
GstBuffer *buf;
guint tdiff;
src = GST_AUDIOTESTSRC (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, "audiotest 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_audiotestsrc_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);
src->timestamp += tdiff;
src->offset += src->samples_per_buffer;
src->process (src, (gint16 *) GST_BUFFER_DATA (buf));
*buffer = buf;
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_audiotestsrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioTestSrc *src = GST_AUDIOTESTSRC (object);
switch (prop_id) {
case PROP_SAMPLES_PER_BUFFER:
src->samples_per_buffer = g_value_get_int (value);
break;
case PROP_WAVE:
src->wave = g_value_get_enum (value);
gst_audiotestsrc_change_wave (src);
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_audiotestsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioTestSrc *src = GST_AUDIOTESTSRC (object);
switch (prop_id) {
case PROP_SAMPLES_PER_BUFFER:
g_value_set_int (value, src->samples_per_buffer);
break;
case PROP_WAVE:
g_value_set_enum (value, src->wave);
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_audiotestsrc_start (GstBaseSrc * basesrc)
{
GstAudioTestSrc *src = GST_AUDIOTESTSRC (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, "audiotestsrc",
GST_RANK_NONE, GST_TYPE_AUDIOTESTSRC);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audiotestsrc",
"Creates audio test signals of given frequency and volume",
plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)

93
gst/audiotestsrc/gstaudiotestsrc.h Normal file
View file

@ -0,0 +1,93 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstsinesrc.h:
*
* 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.
*/
#ifndef __GST_AUDIOTESTSRC_H__
#define __GST_AUDIOTESTSRC_H__
#include <gst/gst.h>
#include <gst/base/gstbasesrc.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIOTESTSRC \
(gst_audiotestsrc_get_type())
#define GST_AUDIOTESTSRC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIOTESTSRC,GstAudioTestSrc))
#define GST_AUDIOTESTSRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIOTESTSRC,GstAudioTestSrcClass))
#define GST_IS_AUDIOTESTSRC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIOTESTSRC))
#define GST_IS_AUDIOTESTSRC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOTESTSRC))
typedef enum {
GST_AUDIOTESTSRC_WAVE_SINE,
GST_AUDIOTESTSRC_WAVE_SQUARE,
GST_AUDIOTESTSRC_WAVE_SAW,
GST_AUDIOTESTSRC_WAVE_TRIANGLE,
GST_AUDIOTESTSRC_WAVE_SILENCE,
GST_AUDIOTESTSRC_WAVE_WHITE_NOISE,
} GstAudioTestSrcWaves;
typedef struct _GstAudioTestSrc GstAudioTestSrc;
typedef struct _GstAudioTestSrcClass GstAudioTestSrcClass;
struct _GstAudioTestSrc {
GstBaseSrc parent;
void (*process)(GstAudioTestSrc*, gint16 *);
/* parameters */
GstAudioTestSrcWaves wave;
gdouble volume;
gdouble freq;
/* audio parameters */
gint samplerate;
gint samples_per_buffer;
guint64 timestamp;
guint64 offset;
gdouble accumulator;
gboolean tags_pushed;
GstClockID clock_id;
GstClockTimeDiff timestamp_offset;
};
struct _GstAudioTestSrcClass {
GstBaseSrcClass parent_class;
};
GType gst_audiotestsrc_get_type(void);
gboolean gst_audiotestsrc_factory_init (GstElementFactory *factory);
G_END_DECLS
#endif /* __GST_AUDIOTESTSRC_H__ */

2
gst/sine/gstsinesrc.c
View file

@ -106,7 +106,7 @@ gst_sinesrc_class_init (GstSineSrcClass * klass)
gobject_class->set_property = gst_sinesrc_set_property;
gobject_class->get_property = gst_sinesrc_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass),
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",