gstreamer/gst/adder/gstadder.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Thomas <thomas@apestaart.org>
* 2005,2006 Wim Taymans <wim@fluendo.com>
*
* 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.
*/
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
/**
* SECTION:element-adder
*
* <refsect2>
* <para>
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* The Adder allows to mix several streams into one by adding the data.
* Mixed data is clamped to the min/max values of the data format.
* </para>
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=100 ! adder name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
* </programlisting>
* This pipeline produces two sine waves mixed together.
* </para>
* <para>
* The Adder currently mixes all data received on the sinkpads as soon as possible
* without trying to synchronize the streams.
* </para>
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* </refsect2>
*
* Last reviewed on 2006-05-09 (0.10.7)
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
*/
/* 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_32 ((gint32) (0x7fffffff))
#define MAX_INT_16 ((gint16) (0x7fff))
#define MAX_INT_8 ((gint8) (0x7f))
#define MAX_UINT_32 ((guint32)(0xffffffff))
#define MAX_UINT_16 ((guint16)(0xffff))
#define MAX_UINT_8 ((guint8) (0xff))
#define MIN_INT_32 ((gint32) (0x80000000))
#define MIN_INT_16 ((gint16) (0x8000))
#define MIN_INT_8 ((gint8) (0x80))
#define MIN_UINT_32 ((guint32)(0x00000000))
#define MIN_UINT_16 ((guint16)(0x0000))
#define MIN_UINT_8 ((guint8) (0x00))
#define GST_CAT_DEFAULT gst_adder_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* elementfactory information */
#define CAPS \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"depth = (int) 32, " \
"signed = (boolean) { true, false } ;" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) { true, false } ;" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 8, " \
"depth = (int) 8, " \
"signed = (boolean) { true, false } ;" \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) { 32, 64 }"
static GstStaticPadTemplate gst_adder_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS)
);
static GstStaticPadTemplate gst_adder_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS (CAPS)
);
static void gst_adder_class_init (GstAdderClass * klass);
static void gst_adder_init (GstAdder * adder);
static void gst_adder_finalize (GObject * object);
static gboolean gst_adder_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_adder_query (GstPad * pad, GstQuery * query);
static gboolean gst_adder_src_event (GstPad * pad, GstEvent * event);
static gboolean gst_adder_sink_event (GstPad * pad, GstEvent * event);
static GstPad *gst_adder_request_new_pad (GstElement * element,
GstPadTemplate * temp, const gchar * unused);
static void gst_adder_release_pad (GstElement * element, GstPad * pad);
static GstStateChangeReturn gst_adder_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_adder_collected (GstCollectPads * pads,
gpointer user_data);
static GstElementClass *parent_class = NULL;
GType
gst_adder_get_type (void)
{
static GType adder_type = 0;
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
if (G_UNLIKELY (adder_type == 0)) {
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_CAT_DEFAULT, "adder", 0,
"audio channel mixing element");
}
return adder_type;
}
/* clipping versions */
#define MAKE_FUNC(name,type,ttype,min,max) \
static void name (type *out, type *in, gint bytes) { \
gint i; \
for (i = 0; i < bytes / sizeof (type); i++) \
out[i] = CLAMP ((ttype)out[i] + (ttype)in[i], min, max); \
}
/* non-clipping versions (for float) */
#define MAKE_FUNC_NC(name,type,ttype) \
static void name (type *out, type *in, gint bytes) { \
gint i; \
for (i = 0; i < bytes / sizeof (type); i++) \
out[i] = (ttype)out[i] + (ttype)in[i]; \
}
/* *INDENT-OFF* */
MAKE_FUNC (add_int32, gint32, gint64, MIN_INT_32, MAX_INT_32)
MAKE_FUNC (add_int16, gint16, gint32, MIN_INT_16, MAX_INT_16)
MAKE_FUNC (add_int8, gint8, gint16, MIN_INT_8, MAX_INT_8)
MAKE_FUNC (add_uint32, guint32, guint64, MIN_UINT_32, MAX_UINT_32)
MAKE_FUNC (add_uint16, guint16, guint32, MIN_UINT_16, MAX_UINT_16)
MAKE_FUNC (add_uint8, guint8, guint16, MIN_UINT_8, MAX_UINT_8)
MAKE_FUNC_NC (add_float64, gdouble, gdouble)
MAKE_FUNC_NC (add_float32, gfloat, gfloat)
/* *INDENT-ON* */
/* we can only accept caps that we and downstream can handle. */
static GstCaps *
gst_adder_sink_getcaps (GstPad * pad)
{
GstAdder *adder;
GstCaps *result, *peercaps, *sinkcaps;
adder = GST_ADDER (GST_PAD_PARENT (pad));
GST_OBJECT_LOCK (adder);
/* get the downstream possible caps */
peercaps = gst_pad_peer_get_caps (adder->srcpad);
/* get the allowed caps on this sinkpad, we use the fixed caps function so
* that it does not call recursively in this function. */
sinkcaps = gst_pad_get_fixed_caps_func (pad);
if (peercaps) {
/* if the peer has caps, intersect */
GST_DEBUG_OBJECT (adder, "intersecting peer and template caps");
result = gst_caps_intersect (peercaps, sinkcaps);
gst_caps_unref (peercaps);
gst_caps_unref (sinkcaps);
} else {
/* the peer has no caps (or there is no peer), just use the allowed caps
* of this sinkpad. */
GST_DEBUG_OBJECT (adder, "no peer caps, using sinkcaps");
result = sinkcaps;
}
GST_OBJECT_UNLOCK (adder);
return result;
}
/* the first caps we receive on any of the sinkpads will define the caps for all
* the other sinkpads because we can only mix streams with the same caps.
* */
static gboolean
gst_adder_setcaps (GstPad * pad, GstCaps * caps)
{
GstAdder *adder;
GList *pads;
GstStructure *structure;
const char *media_type;
adder = GST_ADDER (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (adder, "setting caps on pad %p,%s to %" GST_PTR_FORMAT, pad,
GST_PAD_NAME (pad), caps);
/* FIXME, see if the other pads can accept the format. Also lock the
* format on the other pads to this new format. */
GST_OBJECT_LOCK (adder);
pads = GST_ELEMENT (adder)->pads;
while (pads) {
GstPad *otherpad = GST_PAD (pads->data);
if (otherpad != pad) {
gst_caps_replace (&GST_PAD_CAPS (otherpad), caps);
}
pads = g_list_next (pads);
}
GST_OBJECT_UNLOCK (adder);
/* parse caps now */
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_OBJECT (adder, "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);
if (adder->endianness != G_BYTE_ORDER)
goto not_supported;
switch (adder->width) {
case 8:
adder->func = (adder->is_signed ?
(GstAdderFunction) add_int8 : (GstAdderFunction) add_uint8);
break;
case 16:
adder->func = (adder->is_signed ?
(GstAdderFunction) add_int16 : (GstAdderFunction) add_uint16);
break;
case 32:
adder->func = (adder->is_signed ?
(GstAdderFunction) add_int32 : (GstAdderFunction) add_uint32);
break;
default:
goto not_supported;
}
} else if (strcmp (media_type, "audio/x-raw-float") == 0) {
GST_DEBUG_OBJECT (adder, "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, "endianness", &adder->endianness);
if (adder->endianness != G_BYTE_ORDER)
goto not_supported;
switch (adder->width) {
case 32:
adder->func = (GstAdderFunction) add_float32;
break;
case 64:
adder->func = (GstAdderFunction) add_float64;
break;
default:
goto not_supported;
}
} else {
goto not_supported;
}
gst_structure_get_int (structure, "channels", &adder->channels);
gst_structure_get_int (structure, "rate", &adder->rate);
/* precalc bps */
adder->bps = (adder->width / 8) * adder->channels;
return TRUE;
/* ERRORS */
not_supported:
{
GST_DEBUG_OBJECT (adder, "unsupported format set as caps");
return FALSE;
}
}
/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* eachother which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_adder_query_duration (GstAdder * adder, GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (!done) {
GstIteratorResult ires;
gpointer item;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = GST_PAD_CAST (item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_query_peer_duration (pad, &format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (adder, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static gboolean
gst_adder_query_latency (GstAdder * adder, GstQuery * query)
{
GstClockTime min, max;
gboolean live;
gboolean res;
GstIterator *it;
gboolean done;
res = TRUE;
done = FALSE;
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
/* Take maximum of all latency values */
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (!done) {
GstIteratorResult ires;
gpointer item;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = GST_PAD_CAST (item);
GstQuery *peerquery;
GstClockTime min_cur, max_cur;
gboolean live_cur;
peerquery = gst_query_new_latency ();
/* Ask peer for latency */
res &= gst_pad_peer_query (pad, peerquery);
/* take max from all valid return values */
if (res) {
gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur);
if (min_cur > min)
min = min_cur;
if (max_cur != GST_CLOCK_TIME_NONE &&
((max != GST_CLOCK_TIME_NONE && max_cur > max) ||
(max == GST_CLOCK_TIME_NONE)))
max = max_cur;
live = live || live_cur;
}
gst_query_unref (peerquery);
break;
}
case GST_ITERATOR_RESYNC:
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
gst_iterator_free (it);
if (res) {
/* store the results */
GST_DEBUG_OBJECT (adder, "Calculated total latency: live %s, min %"
GST_TIME_FORMAT ", max %" GST_TIME_FORMAT,
(live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
return res;
}
static gboolean
gst_adder_query (GstPad * pad, GstQuery * query)
{
GstAdder *adder = GST_ADDER (gst_pad_get_parent (pad));
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
/* FIXME, bring to stream time, might be tricky */
gst_query_set_position (query, format, adder->timestamp);
res = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, adder->offset);
res = TRUE;
break;
default:
break;
}
break;
}
case GST_QUERY_DURATION:
res = gst_adder_query_duration (adder, query);
break;
case GST_QUERY_LATENCY:
res = gst_adder_query_latency (adder, query);
break;
default:
/* FIXME, needs a custom query handler because we have multiple
* sinkpads */
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (adder);
return res;
}
static gboolean
forward_event_func (GstPad * pad, GValue * ret, GstEvent * event)
{
gst_event_ref (event);
GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event));
if (!gst_pad_push_event (pad, event)) {
g_value_set_boolean (ret, FALSE);
GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.",
event, GST_EVENT_TYPE_NAME (event));
} else {
GST_LOG_OBJECT (pad, "Sent event %p (%s).",
event, GST_EVENT_TYPE_NAME (event));
}
gst_object_unref (pad);
return TRUE;
}
/* forwards the event to all sinkpads, takes ownership of the
* event
*
* Returns: TRUE if the event could be forwarded on all
* sinkpads.
*/
static gboolean
forward_event (GstAdder * adder, GstEvent * event)
{
gboolean ret;
GstIterator *it;
GValue vret = { 0 };
GST_LOG_OBJECT (adder, "Forwarding event %p (%s)", event,
GST_EVENT_TYPE_NAME (event));
ret = TRUE;
g_value_init (&vret, G_TYPE_BOOLEAN);
g_value_set_boolean (&vret, TRUE);
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func, &vret,
event);
gst_iterator_free (it);
gst_event_unref (event);
ret = g_value_get_boolean (&vret);
return ret;
}
static gboolean
gst_adder_src_event (GstPad * pad, GstEvent * event)
{
GstAdder *adder;
gboolean result;
adder = GST_ADDER (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_QOS:
/* QoS might be tricky */
result = FALSE;
break;
case GST_EVENT_SEEK:
{
GstSeekFlags flags;
GstSeekType curtype;
gint64 cur;
/* parse the seek parameters */
gst_event_parse_seek (event, &adder->segment_rate, NULL, &flags, &curtype,
&cur, NULL, NULL);
/* check if we are flushing */
if (flags & GST_SEEK_FLAG_FLUSH) {
/* make sure we accept nothing anymore and return WRONG_STATE */
gst_collect_pads_set_flushing (adder->collect, TRUE);
/* flushing seek, start flush downstream, the flush will be done
* when all pads received a FLUSH_STOP. */
gst_pad_push_event (adder->srcpad, gst_event_new_flush_start ());
}
/* now wait for the collected to be finished and mark a new
* segment */
GST_OBJECT_LOCK (adder->collect);
if (curtype == GST_SEEK_TYPE_SET)
adder->segment_position = cur;
else
adder->segment_position = 0;
adder->segment_pending = TRUE;
GST_OBJECT_UNLOCK (adder->collect);
result = forward_event (adder, event);
break;
}
case GST_EVENT_NAVIGATION:
/* navigation is rather pointless. */
result = FALSE;
break;
default:
/* just forward the rest for now */
result = forward_event (adder, event);
break;
}
gst_object_unref (adder);
return result;
}
static gboolean
gst_adder_sink_event (GstPad * pad, GstEvent * event)
{
GstAdder *adder;
gboolean ret;
adder = GST_ADDER (gst_pad_get_parent (pad));
GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event),
GST_DEBUG_PAD_NAME (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
/* mark a pending new segment. This event is synchronized
* with the streaming thread so we can safely update the
* variable without races. It's somewhat weird because we
* assume the collectpads forwarded the FLUSH_STOP past us
* and downstream (using our source pad, the bastard!).
*/
adder->segment_pending = TRUE;
break;
default:
break;
}
/* now GstCollectPads can take care of the rest, e.g. EOS */
ret = adder->collect_event (pad, event);
gst_object_unref (adder);
return ret;
}
static void
gst_adder_class_init (GstAdderClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = gst_adder_finalize;
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_simple (gstelement_class, "Adder",
"Generic/Audio",
"Add N audio channels together", "Thomas <thomas@apestaart.org>");
parent_class = g_type_class_peek_parent (klass);
gstelement_class->request_new_pad = gst_adder_request_new_pad;
gstelement_class->release_pad = gst_adder_release_pad;
gstelement_class->change_state = gst_adder_change_state;
}
static void
gst_adder_init (GstAdder * adder)
{
GstPadTemplate *template;
template = gst_static_pad_template_get (&gst_adder_src_template);
adder->srcpad = gst_pad_new_from_template (template, "src");
gst_object_unref (template);
gst_pad_set_getcaps_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
gst_pad_set_setcaps_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_setcaps));
gst_pad_set_query_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_query));
gst_pad_set_event_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_src_event));
gst_element_add_pad (GST_ELEMENT (adder), adder->srcpad);
adder->format = GST_ADDER_FORMAT_UNSET;
adder->padcount = 0;
adder->func = NULL;
/* keep track of the sinkpads requested */
adder->collect = gst_collect_pads_new ();
gst_collect_pads_set_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_collected), adder);
}
static void
gst_adder_finalize (GObject * object)
{
GstAdder *adder = GST_ADDER (object);
gst_object_unref (adder->collect);
adder->collect = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstPad *
gst_adder_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * unused)
{
gchar *name;
GstAdder *adder;
GstPad *newpad;
gint padcount;
if (templ->direction != GST_PAD_SINK)
goto not_sink;
adder = GST_ADDER (element);
/* increment pad counter */
padcount = g_atomic_int_exchange_and_add (&adder->padcount, 1);
name = g_strdup_printf ("sink%d", padcount);
newpad = gst_pad_new_from_template (templ, name);
GST_DEBUG_OBJECT (adder, "request new pad %s", name);
g_free (name);
gst_pad_set_getcaps_function (newpad,
GST_DEBUG_FUNCPTR (gst_adder_sink_getcaps));
gst_pad_set_setcaps_function (newpad, GST_DEBUG_FUNCPTR (gst_adder_setcaps));
gst_collect_pads_add_pad (adder->collect, newpad, sizeof (GstCollectData));
/* FIXME: hacked way to override/extend the event function of
* GstCollectPads; because it sets its own event function giving the
* element no access to events */
adder->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad);
gst_pad_set_event_function (newpad, GST_DEBUG_FUNCPTR (gst_adder_sink_event));
/* takes ownership of the pad */
if (!gst_element_add_pad (GST_ELEMENT (adder), newpad))
goto could_not_add;
return newpad;
/* errors */
not_sink:
{
g_warning ("gstadder: request new pad that is not a SINK pad\n");
return NULL;
}
could_not_add:
{
GST_DEBUG_OBJECT (adder, "could not add pad");
gst_collect_pads_remove_pad (adder->collect, newpad);
gst_object_unref (newpad);
return NULL;
}
}
static void
gst_adder_release_pad (GstElement * element, GstPad * pad)
{
GstAdder *adder;
adder = GST_ADDER (element);
GST_DEBUG_OBJECT (adder, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
gst_collect_pads_remove_pad (adder->collect, pad);
gst_element_remove_pad (element, pad);
}
static GstFlowReturn
gst_adder_collected (GstCollectPads * pads, gpointer user_data)
{
/*
* combine channels by adding sample values
* basic algorithm :
* - this function is called when all pads have a buffer
* - get available bytes on all pads.
* - repeat for each input pad :
* - read available bytes, copy or add to target buffer
* - if there's an EOS event, remove the input channel
* - push out the output buffer
*/
GstAdder *adder;
guint size;
GSList *collected;
GstBuffer *outbuf;
GstFlowReturn ret;
gpointer outbytes;
gboolean empty = TRUE;
adder = GST_ADDER (user_data);
/* this is fatal */
if (G_UNLIKELY (adder->func == NULL))
goto not_negotiated;
/* get available bytes for reading, this can be 0 which could mean
* empty buffers or EOS, which we will catch when we loop over the
* pads. */
size = gst_collect_pads_available (pads);
GST_LOG_OBJECT (adder,
"starting to cycle through channels, %d bytes available (bps = %d)", size,
adder->bps);
outbuf = NULL;
outbytes = NULL;
for (collected = pads->data; collected; collected = g_slist_next (collected)) {
GstCollectData *data;
guint8 *bytes;
guint len;
GstBuffer *inbuf;
data = (GstCollectData *) collected->data;
/* get a subbuffer of size bytes */
inbuf = gst_collect_pads_take_buffer (pads, data, size);
/* NULL means EOS or an empty buffer so we still need to flush in
* case of an empty buffer. */
if (inbuf == NULL) {
GST_LOG_OBJECT (adder, "channel %p: no bytes available", data);
goto next;
}
bytes = GST_BUFFER_DATA (inbuf);
len = GST_BUFFER_SIZE (inbuf);
if (outbuf == NULL) {
GST_LOG_OBJECT (adder, "channel %p: making output buffer of %d bytes",
data, size);
/* first buffer, alloc size bytes. FIXME, we can easily subbuffer
* and _make_writable. */
outbuf = gst_buffer_new_and_alloc (size);
outbytes = GST_BUFFER_DATA (outbuf);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (adder->srcpad));
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
/* clear if we are only going to fill a partial buffer */
if (G_UNLIKELY (size > len))
memset (outbytes, 0, size);
GST_LOG_OBJECT (adder, "channel %p: copying %d bytes from data %p",
data, len, bytes);
/* and copy the data into it */
memcpy (outbytes, bytes, len);
empty = FALSE;
} else {
GST_LOG_OBJECT (adder, "channel %p: zeroing %d bytes from data %p",
data, len, bytes);
memset (outbytes, 0, size);
}
} else {
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
GST_LOG_OBJECT (adder, "channel %p: mixing %d bytes from data %p",
data, len, bytes);
/* other buffers, need to add them */
adder->func ((gpointer) outbytes, (gpointer) bytes, len);
empty = FALSE;
} else {
GST_LOG_OBJECT (adder, "channel %p: skipping %d bytes from data %p",
data, len, bytes);
}
}
next:
if (inbuf)
gst_buffer_unref (inbuf);
}
/* can only happen when no pads to collect or all EOS */
if (outbuf == NULL)
goto eos;
/* our timestamping is very simple, just an ever incrementing
* counter, the new segment time will take care of their respective
* stream time. */
if (adder->segment_pending) {
GstEvent *event;
/* FIXME, use rate/applied_rate as set on all sinkpads.
* - currently we just set rate as received from last seek-event
* We could potentially figure out the duration as well using
* the current segment positions and the stated stop positions.
* Also we just start from stream time 0 which is rather
* weird. For non-synchronized mixing, the time should be
* the min of the stream times of all received segments,
* rationale being that the duration is at least going to
* be as long as the earliest stream we start mixing. This
* would also be correct for synchronized mixing but then
* the later streams would be delayed until the stream times
* match.
*/
event = gst_event_new_new_segment_full (FALSE, adder->segment_rate,
1.0, GST_FORMAT_TIME, adder->timestamp, -1, adder->segment_position);
gst_pad_push_event (adder->srcpad, event);
adder->segment_pending = FALSE;
adder->segment_position = 0;
}
/* set timestamps on the output buffer */
GST_BUFFER_TIMESTAMP (outbuf) = adder->timestamp;
GST_BUFFER_OFFSET (outbuf) = adder->offset;
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */
adder->offset += size / adder->bps;
adder->timestamp = gst_util_uint64_scale_int (adder->offset,
GST_SECOND, adder->rate);
/* now we can set the duration of the buffer */
GST_BUFFER_DURATION (outbuf) = adder->timestamp -
GST_BUFFER_TIMESTAMP (outbuf);
/* if we only processed silence, mark output again as silence */
if (empty)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
/* send it out */
GST_LOG_OBJECT (adder, "pushing outbuf, timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (adder->srcpad, outbuf);
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
eos:
{
GST_DEBUG_OBJECT (adder, "no data available, must be EOS");
gst_pad_push_event (adder->srcpad, gst_event_new_eos ());
return GST_FLOW_UNEXPECTED;
}
}
static GstStateChangeReturn
gst_adder_change_state (GstElement * element, GstStateChange transition)
{
GstAdder *adder;
GstStateChangeReturn ret;
adder = GST_ADDER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
adder->timestamp = 0;
adder->offset = 0;
adder->segment_pending = TRUE;
adder->segment_position = 0;
adder->segment_rate = 1.0;
gst_segment_init (&adder->segment, GST_FORMAT_UNDEFINED);
gst_collect_pads_start (adder->collect);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* need to unblock the collectpads before calling the
* parent change_state so that streaming can finish */
gst_collect_pads_stop (adder->collect);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
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_NAME, GST_PACKAGE_ORIGIN)