gstreamer/gst/adder/gstadder.c
David Schleef 129c7e8af1 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

531 lines
16 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wim.taymans@chello.be>
* 2001 Thomas <thomas@apestaart.org>
*
* adder.c: Adder element, N in, one out, samples are added
*
* 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.
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstadder.h"
#include <gst/audio/audio.h>
#include <string.h> /* strcmp */
/* highest positive/lowest negative x-bit value we can use for clamping */
//#define MAX_INT_x ((guint) 1 << (x - 1))
#define MAX_INT_32 2147483647L
#define MAX_INT_16 32767
#define MAX_INT_8 127
//#define MIN_INT_x (-((guint) 1 << (x - 1)))
/* to make non-C90 happy we need to specify the constant differently */
#define MIN_INT_32 (-2147483647L -1L)
#define MIN_INT_16 -32768
#define MIN_INT_8 -128
#define GST_ADDER_BUFFER_SIZE 4096
#define GST_ADDER_NUM_BUFFERS 8
GST_DEBUG_CATEGORY_STATIC (gst_adder_debug);
#define GST_CAT_DEFAULT gst_adder_debug
/* elementfactory information */
static GstElementDetails adder_details = GST_ELEMENT_DETAILS ("Adder",
"Generic/Audio",
"Add N audio channels together",
"Thomas <thomas@apestaart.org>");
/* Adder signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_NUM_PADS
/* FILL ME */
};
static GstStaticPadTemplate gst_adder_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
);
static GstStaticPadTemplate gst_adder_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
);
static void gst_adder_class_init (GstAdderClass * klass);
static void gst_adder_init (GstAdder * adder);
static void gst_adder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstPad *gst_adder_request_new_pad (GstElement * element,
GstPadTemplate * temp, const gchar * unused);
static GstElementStateReturn gst_adder_change_state (GstElement * element);
/* we do need a loop function */
static void gst_adder_loop (GstElement * element);
static GstElementClass *parent_class = NULL;
/* static guint gst_adder_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_adder_get_type (void)
{
static GType adder_type = 0;
if (!adder_type) {
static const GTypeInfo adder_info = {
sizeof (GstAdderClass), NULL, NULL,
(GClassInitFunc) gst_adder_class_init, NULL, NULL,
sizeof (GstAdder), 0,
(GInstanceInitFunc) gst_adder_init,
};
adder_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAdder",
&adder_info, 0);
GST_DEBUG_CATEGORY_INIT (gst_adder_debug, "adder", 0,
"audio channel mixing element");
}
return adder_type;
}
static GstPadLinkReturn
gst_adder_link (GstPad * pad, const GstCaps * caps)
{
GstAdder *adder;
const char *media_type;
const GList *pads;
GstStructure *structure;
GstPadLinkReturn ret;
GstElement *element;
g_return_val_if_fail (caps != NULL, GST_PAD_LINK_REFUSED);
g_return_val_if_fail (pad != NULL, GST_PAD_LINK_REFUSED);
element = GST_PAD_PARENT (pad);
adder = GST_ADDER (element);
pads = gst_element_get_pad_list (element);
while (pads) {
GstPad *otherpad = GST_PAD (pads->data);
if (otherpad != pad) {
ret = gst_pad_try_set_caps (otherpad, caps);
if (GST_PAD_LINK_FAILED (ret)) {
return ret;
}
}
pads = g_list_next (pads);
}
pads = gst_element_get_pad_list (GST_ELEMENT (adder));
while (pads) {
GstPad *otherpad = GST_PAD (pads->data);
if (otherpad != pad) {
ret = gst_pad_try_set_caps (otherpad, caps);
if (GST_PAD_LINK_FAILED (ret)) {
return ret;
}
}
pads = g_list_next (pads);
}
structure = gst_caps_get_structure (caps, 0);
media_type = gst_structure_get_name (structure);
if (strcmp (media_type, "audio/x-raw-int") == 0) {
GST_DEBUG ("parse_caps sets adder to format int");
adder->format = GST_ADDER_FORMAT_INT;
gst_structure_get_int (structure, "width", &adder->width);
gst_structure_get_int (structure, "depth", &adder->depth);
gst_structure_get_int (structure, "endianness", &adder->endianness);
gst_structure_get_boolean (structure, "signed", &adder->is_signed);
gst_structure_get_int (structure, "channels", &adder->channels);
gst_structure_get_int (structure, "rate", &adder->rate);
} else if (strcmp (media_type, "audio/x-raw-float") == 0) {
GST_DEBUG ("parse_caps sets adder to format float");
adder->format = GST_ADDER_FORMAT_FLOAT;
gst_structure_get_int (structure, "width", &adder->width);
gst_structure_get_int (structure, "channels", &adder->channels);
gst_structure_get_int (structure, "rate", &adder->rate);
}
return GST_PAD_LINK_OK;
}
static void
gst_adder_class_init (GstAdderClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_adder_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_adder_sink_template));
gst_element_class_set_details (gstelement_class, &adder_details);
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_NUM_PADS,
g_param_spec_int ("num_pads", "number of pads", "Number Of Pads",
0, G_MAXINT, 0, G_PARAM_READABLE));
gobject_class->get_property = gst_adder_get_property;
gstelement_class->request_new_pad = gst_adder_request_new_pad;
gstelement_class->change_state = gst_adder_change_state;
}
static void
gst_adder_init (GstAdder * adder)
{
adder->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_adder_src_template), "src");
gst_element_add_pad (GST_ELEMENT (adder), adder->srcpad);
gst_element_set_loop_function (GST_ELEMENT (adder), gst_adder_loop);
gst_pad_set_getcaps_function (adder->srcpad, gst_pad_proxy_getcaps);
gst_pad_set_link_function (adder->srcpad, gst_adder_link);
adder->format = GST_ADDER_FORMAT_UNSET;
/* keep track of the sinkpads requested */
adder->numsinkpads = 0;
adder->input_channels = NULL;
}
static GstPad *
gst_adder_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * unused)
{
gchar *name;
GstAdder *adder;
GstAdderInputChannel *input;
g_return_val_if_fail (GST_IS_ADDER (element), NULL);
if (templ->direction != GST_PAD_SINK) {
g_warning ("gstadder: request new pad that is not a SINK pad\n");
return NULL;
}
/* allocate space for the input_channel */
input = (GstAdderInputChannel *) g_malloc (sizeof (GstAdderInputChannel));
if (input == NULL) {
g_warning ("gstadder: could not allocate adder input channel !\n");
return NULL;
}
adder = GST_ADDER (element);
/* fill in input_channel structure */
name = g_strdup_printf ("sink%d", adder->numsinkpads);
input->sinkpad = gst_pad_new_from_template (templ, name);
input->bytestream = gst_bytestream_new (input->sinkpad);
gst_element_add_pad (GST_ELEMENT (adder), input->sinkpad);
gst_pad_set_getcaps_function (input->sinkpad, gst_pad_proxy_getcaps);
gst_pad_set_link_function (input->sinkpad, gst_adder_link);
/* add the input_channel to the list of input channels */
adder->input_channels = g_slist_append (adder->input_channels, input);
adder->numsinkpads++;
return input->sinkpad;
}
static void
gst_adder_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAdder *adder;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_ADDER (object));
adder = GST_ADDER (object);
switch (prop_id) {
case ARG_NUM_PADS:
g_value_set_int (value, adder->numsinkpads);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* use this loop */
static void
gst_adder_loop (GstElement * element)
{
/*
* combine channels by adding sample values
* basic algorithm :
* - request an output buffer from the pool
* - repeat for each input pipe :
* - get number of bytes from the channel's bytestream to fill output buffer
* - if there's an EOS event, remove the input channel
* - otherwise add the gotten bytes to the output buffer
* - push out the output buffer
*/
GstAdder *adder;
GstBuffer *buf_out;
GSList *inputs;
register guint i;
g_return_if_fail (element != NULL);
g_return_if_fail (GST_IS_ADDER (element));
adder = GST_ADDER (element);
/* get new output buffer */
/* FIXME the 1024 is arbitrary */
buf_out = gst_buffer_new_and_alloc (1024);
if (buf_out == NULL) {
GST_ELEMENT_ERROR (adder, CORE, TOO_LAZY, (NULL),
("could not get new output buffer"));
return;
}
/* initialize the output data to 0 */
memset (GST_BUFFER_DATA (buf_out), 0, GST_BUFFER_SIZE (buf_out));
/* get data from all of the sinks */
inputs = adder->input_channels;
GST_LOG ("starting to cycle through channels");
while (inputs) {
guint32 got_bytes;
guint8 *raw_in;
GstAdderInputChannel *input;
input = (GstAdderInputChannel *) inputs->data;
inputs = inputs->next;
GST_LOG_OBJECT (adder, " looking into channel %p", input);
if (!GST_PAD_IS_USABLE (input->sinkpad)) {
GST_LOG_OBJECT (adder, " adder ignoring pad %s:%s",
GST_DEBUG_PAD_NAME (input->sinkpad));
continue;
}
/* Get data from the bytestream of each input channel. We need to check for
events before passing on the data to the output buffer. */
repeat:
got_bytes = gst_bytestream_peek_bytes (input->bytestream, &raw_in,
GST_BUFFER_SIZE (buf_out));
/* FIXME we should do something with the data if got_bytes > 0 */
if (got_bytes < GST_BUFFER_SIZE (buf_out)) {
GstEvent *event = NULL;
guint32 waiting;
/* we need to check for an event. */
gst_bytestream_get_status (input->bytestream, &waiting, &event);
if (event) {
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* if we get an EOS event from one of our sink pads, we assume that
pad's finished handling data. just skip this pad. */
GST_DEBUG (" got an EOS event");
gst_event_unref (event);
continue;
case GST_EVENT_INTERRUPT:
gst_event_unref (event);
GST_DEBUG (" got an interrupt event");
/* we have to call interrupt here, the scheduler will switch out
this element ASAP or returns TRUE if we need to exit the loop */
if (gst_element_interrupt (GST_ELEMENT (adder))) {
gst_buffer_unref (buf_out);
return;
}
default:
GST_LOG_OBJECT (adder, "pulling again after event");
goto repeat;
}
}
} else {
/* here's where the data gets copied. */
GST_LOG (" copying %d bytes (format %d,%d)",
GST_BUFFER_SIZE (buf_out), adder->format, adder->width);
GST_LOG (" from channel %p from input data %p", input, raw_in);
GST_LOG (" to output data %p in buffer %p",
GST_BUFFER_DATA (buf_out), buf_out);
if (adder->format == GST_ADDER_FORMAT_INT) {
if (adder->width == 32) {
gint32 *in = (gint32 *) raw_in;
gint32 *out = (gint32 *) GST_BUFFER_DATA (buf_out);
for (i = 0; i < GST_BUFFER_SIZE (buf_out) / 4; i++)
out[i] = CLAMP (((gint64) out[i]) + ((gint64) in[i]),
MIN_INT_32, MAX_INT_32);
} else if (adder->width == 16) {
gint16 *in = (gint16 *) raw_in;
gint16 *out = (gint16 *) GST_BUFFER_DATA (buf_out);
for (i = 0; i < GST_BUFFER_SIZE (buf_out) / 2; i++)
out[i] = CLAMP (out[i] + in[i], MIN_INT_16, MAX_INT_16);
} else if (adder->width == 8) {
gint8 *in = (gint8 *) raw_in;
gint8 *out = (gint8 *) GST_BUFFER_DATA (buf_out);
for (i = 0; i < GST_BUFFER_SIZE (buf_out); i++)
out[i] = CLAMP (out[i] + in[i], MIN_INT_8, MAX_INT_8);
} else {
GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL),
("invalid width (%u) for integer audio in gstadder",
adder->width));
return;
}
} else if (adder->format == GST_ADDER_FORMAT_FLOAT) {
if (adder->width == 64) {
gdouble *in = (gdouble *) raw_in;
gdouble *out = (gdouble *) GST_BUFFER_DATA (buf_out);
for (i = 0; i < GST_BUFFER_SIZE (buf_out) / sizeof (gdouble); i++)
out[i] = CLAMP (out[i] + in[i], -1.0, 1.0);
} else if (adder->width == 32) {
gfloat *in = (gfloat *) raw_in;
gfloat *out = (gfloat *) GST_BUFFER_DATA (buf_out);
for (i = 0; i < GST_BUFFER_SIZE (buf_out) / sizeof (gfloat); i++)
out[i] = CLAMP (out[i] + in[i], -1.0, 1.0);
} else {
GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL),
("invalid width (%u) for float audio in gstadder", adder->width));
return;
}
} else {
GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL),
("invalid audio format (%d) in gstadder", adder->format));
return;
}
gst_bytestream_flush (input->bytestream, GST_BUFFER_SIZE (buf_out));
GST_LOG ("done copying data");
}
}
if (adder->width == 0) {
GST_ELEMENT_ERROR (adder, CORE, NEGOTIATION, (NULL), ("width is 0"));
return;
}
if (adder->channels == 0) {
GST_ELEMENT_ERROR (adder, CORE, NEGOTIATION, (NULL), ("channels is 0"));
return;
}
if (adder->rate == 0) {
GST_ELEMENT_ERROR (adder, CORE, NEGOTIATION, (NULL), ("rate is 0"));
return;
}
GST_BUFFER_TIMESTAMP (buf_out) = adder->timestamp;
if (adder->format == GST_ADDER_FORMAT_FLOAT)
adder->offset += GST_BUFFER_SIZE (buf_out) / adder->width / adder->channels;
else
adder->offset +=
GST_BUFFER_SIZE (buf_out) * 8 / adder->width / adder->channels;
adder->timestamp = adder->offset * GST_SECOND / adder->rate;
/* send it out */
GST_LOG ("pushing buf_out");
gst_pad_push (adder->srcpad, GST_DATA (buf_out));
}
static GstElementStateReturn
gst_adder_change_state (GstElement * element)
{
GstAdder *adder;
adder = GST_ADDER (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
break;
case GST_STATE_READY_TO_PAUSED:
adder->timestamp = 0;
adder->offset = 0;
break;
case GST_STATE_PAUSED_TO_PLAYING:
case GST_STATE_PLAYING_TO_PAUSED:
case GST_STATE_PAUSED_TO_READY:
case GST_STATE_READY_TO_NULL:
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "adder", GST_RANK_NONE, GST_TYPE_ADDER)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"adder",
"Adds multiple streams",
plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)