gstreamer/ext/dts/gstdtsdec.c
Sebastian Dröge 59c9a6a6bd Add support for building against libdca (with the libdts compat header). Fixes bug #511530.
Original commit message from CVS:
* configure.ac:
* ext/dts/gstdtsdec.c:
Add support for building against libdca (with the libdts compat
header). Fixes bug #511530.
Should probably be ported to libdca as some points as it's the
successor of libdts.
2008-01-23 15:51:20 +00:00

740 lines
20 KiB
C

/* GStreamer DTS decoder plugin based on libdtsdec
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* 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.
*/
/* TODO: - Port to libdca API instead of relying on the compat header.
* libdca is the successor of libdts:
* http://www.videolan.org/developers/libdca.html
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "_stdint.h"
#include <stdlib.h>
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <dts.h>
#include "gstdtsdec.h"
#include <liboil/liboil.h>
#include <liboil/liboilcpu.h>
#include <liboil/liboilfunction.h>
GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
#define GST_CAT_DEFAULT (dtsdec_debug)
static const GstElementDetails gst_dtsdec_details =
GST_ELEMENT_DETAILS ("DTS audio decoder",
"Codec/Decoder/Audio",
"Decodes DTS audio streams",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
enum
{
ARG_0,
ARG_DRC
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-dts;" "audio/x-private1-dts")
);
#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
#define DTS_CAPS "audio/x-raw-int, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"signed = (boolean) true, " \
"width = (int) 16, " \
"depth = (int) 16"
#define SAMPLE_WIDTH 16
#elif defined(LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"width = (int) 64"
#define SAMPLE_WIDTH 64
#else
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"width = (int) 32"
#define SAMPLE_WIDTH 32
#endif
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (DTS_CAPS ", "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT);
static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element,
GstStateChange transition);
static void gst_dtsdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtsdec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_dtsdec_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 (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details (element_class, &gst_dtsdec_details);
GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
}
static void
gst_dtsdec_class_init (GstDtsDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_dtsdec_set_property;
gobject_class->get_property = gst_dtsdec_get_property;
gstelement_class->change_state = gst_dtsdec_change_state;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
g_param_spec_boolean ("drc", "Dynamic Range Compression",
"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
oil_init ();
klass->dts_cpuflags = 0;
cpuflags = oil_cpu_get_flags ();
if (cpuflags & OIL_IMPL_FLAG_MMX)
klass->dts_cpuflags |= MM_ACCEL_X86_MMX;
if (cpuflags & OIL_IMPL_FLAG_3DNOW)
klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW;
if (cpuflags & OIL_IMPL_FLAG_MMXEXT)
klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT;
GST_LOG ("CPU flags: dts=%08x, liboil=%08x", klass->dts_cpuflags, cpuflags);
}
static void
gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
{
dtsdec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_sink_setcaps));
gst_pad_set_chain_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
gst_pad_set_event_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_sink_event));
gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (dtsdec->srcpad);
gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad);
dtsdec->dynamic_range_compression = FALSE;
}
static gint
gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
{
gint chans = 0;
GstAudioChannelPosition *tpos = NULL;
if (pos) {
/* Allocate the maximum, for ease */
tpos = *pos = g_new (GstAudioChannelPosition, 7);
if (!tpos)
return 0;
}
switch (flags & DTS_CHANNEL_MASK) {
case DTS_MONO:
chans = 1;
if (tpos)
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
break;
/* case DTS_CHANNEL: */
case DTS_STEREO:
case DTS_STEREO_SUMDIFF:
case DTS_STEREO_TOTAL:
case DTS_DOLBY:
chans = 2;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
break;
case DTS_3F:
chans = 3;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
break;
case DTS_2F1R:
chans = 3;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
break;
case DTS_3F1R:
chans = 4;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
break;
case DTS_2F2R:
chans = 4;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
case DTS_3F2R:
chans = 5;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
case DTS_4F2R:
chans = 6;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
default:
g_warning ("dtsdec: invalid flags 0x%x", flags);
return 0;
}
if (flags & DTS_LFE) {
if (tpos) {
tpos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE;
}
chans += 1;
}
return chans;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
GstAudioChannelPosition *pos;
GstCaps *caps = gst_caps_from_string (DTS_CAPS);
gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
gboolean result = FALSE;
if (!channels)
goto done;
GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (dts->srcpad, caps))
goto done;
result = TRUE;
done:
if (caps) {
gst_caps_unref (caps);
}
return result;
}
static gboolean
gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
{
GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad));
gboolean ret = FALSE;
GST_LOG_OBJECT (dtsdec, "%s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat format;
gint64 val;
gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL,
NULL);
if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) {
GST_WARNING ("No time in newsegment event %p", event);
} else {
dtsdec->current_ts = val;
}
if (dtsdec->cache) {
gst_buffer_unref (dtsdec->cache);
dtsdec->cache = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
}
case GST_EVENT_FLUSH_STOP:
if (dtsdec->cache) {
gst_buffer_unref (dtsdec->cache);
dtsdec->cache = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (dtsdec);
return ret;
}
static gboolean
gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
dts->dvdmode = TRUE;
else
dts->dvdmode = FALSE;
gst_object_unref (dts);
return TRUE;
}
static void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
}
static GstFlowReturn
gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
gboolean need_renegotiation = FALSE;
gint channels, num_blocks;
GstBuffer *out;
gint i, s, c, num_c;
sample_t *samples;
GstFlowReturn result = GST_FLOW_OK;
/* go over stream properties, update caps/streaminfo if needed */
if (dts->sample_rate != sample_rate) {
need_renegotiation = TRUE;
dts->sample_rate = sample_rate;
}
dts->stream_channels = flags;
if (bit_rate != dts->bit_rate) {
dts->bit_rate = bit_rate;
gst_dtsdec_update_streaminfo (dts);
}
if (dts->request_channels == DTS_CHANNEL) {
GstCaps *caps;
caps = gst_pad_get_allowed_caps (dts->srcpad);
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
GstStructure *structure = gst_caps_get_structure (copy, 0);
gint channels;
const int dts_channels[6] = {
DTS_MONO,
DTS_STEREO,
DTS_STEREO | DTS_LFE,
DTS_2F2R,
DTS_2F2R | DTS_LFE,
DTS_3F2R | DTS_LFE,
};
/* Prefer the original number of channels, but fixate to something
* preferred (first in the caps) downstream if possible.
*/
gst_structure_fixate_field_nearest_int (structure, "channels",
flags ? gst_dtsdec_channels (flags, NULL) : 6);
gst_structure_get_int (structure, "channels", &channels);
if (channels <= 6)
dts->request_channels = dts_channels[channels - 1];
else
dts->request_channels = dts_channels[5];
gst_caps_unref (copy);
} else if (flags) {
dts->request_channels = dts->stream_channels;
} else {
dts->request_channels = DTS_3F2R | DTS_LFE;
}
if (caps)
gst_caps_unref (caps);
}
/* process */
flags = dts->request_channels | DTS_ADJUST_LEVEL;
dts->level = 1;
if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
GST_WARNING ("dts_frame error");
return GST_FLOW_OK;
}
channels = flags & (DTS_CHANNEL_MASK | DTS_LFE);
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
dts->using_channels = channels;
}
if (need_renegotiation == TRUE) {
GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
dts->sample_rate, dts->stream_channels, dts->using_channels);
if (!gst_dtsdec_renegotiate (dts)) {
GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
}
if (dts->dynamic_range_compression == FALSE) {
dts_dynrng (dts->state, NULL, NULL);
}
/* handle decoded data, one block is 256 samples */
num_blocks = dts_blocks_num (dts->state);
for (i = 0; i < num_blocks; i++) {
if (dts_block (dts->state)) {
GST_WARNING ("dts_block error %d", i);
continue;
}
samples = dts_samples (dts->state);
num_c = gst_dtsdec_channels (dts->using_channels, NULL);
result = gst_pad_alloc_buffer_and_set_caps (dts->srcpad, 0,
(SAMPLE_WIDTH / 8) * 256 * num_c, GST_PAD_CAPS (dts->srcpad), &out);
if (result != GST_FLOW_OK)
break;
GST_BUFFER_TIMESTAMP (out) = dts->current_ts;
GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
dts->current_ts += GST_BUFFER_DURATION (out);
/* libdts returns buffers in 256-sample-blocks per channel,
* we want interleaved. And we need to copy anyway... */
data = GST_BUFFER_DATA (out);
for (s = 0; s < 256; s++) {
for (c = 0; c < num_c; c++) {
*(sample_t *) data = samples[s + c * 256];
data += (SAMPLE_WIDTH / 8);
}
}
/* push on */
result = gst_pad_push (dts->srcpad, out);
if (result != GST_FLOW_OK)
break;
}
return result;
}
static GstFlowReturn
gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstDtsDec *dts;
guint8 *data;
gint size;
gint length, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
dts = GST_DTSDEC (GST_PAD_PARENT (pad));
if (dts->cache) {
buf = gst_buffer_join (dts->cache, buf);
dts->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
length = 0;
while (size >= 7) {
length = dts_syncinfo (dts->state, data, &flags,
&sample_rate, &bit_rate, &frame_length);
if (length == 0) {
/* shift window to re-find sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: frame size %d", length);
result = gst_dtsdec_handle_frame (dts, data, length,
flags, sample_rate, bit_rate);
if (result != GST_FLOW_OK) {
size = 0;
break;
}
size -= length;
data += length;
} else {
GST_LOG ("Not enough data available (needed %d had %d)", length, size);
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
dts->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
return result;
}
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
gint first_access;
if (dts->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guint8 *data = GST_BUFFER_DATA (buf);
gint offset, len;
GstBuffer *subbuf;
if (size < 2)
goto not_enough_data;
first_access = (data[0] << 8) | data[1];
/* Skip the first_access header */
offset = 2;
if (first_access > 1) {
/* Length of data before first_access */
len = first_access - 1;
if (len <= 0 || offset + len > size)
goto bad_first_access_parameter;
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
res = gst_dtsdec_chain_raw (pad, subbuf);
if (res != GST_FLOW_OK)
goto done;
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
res = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies
* to the first byte */
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
res = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
res = gst_dtsdec_chain_raw (pad, buf);
}
done:
return res;
/* ERRORS */
not_enough_data:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Insufficient data in buffer. Can't determine first_acess"));
return GST_FLOW_ERROR;
}
bad_first_access_parameter:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Bad first_access parameter (%d) in buffer", first_access));
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstDtsDec *dts = GST_DTSDEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstDtsDecClass *klass;
klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
dts->state = dts_init (klass->dts_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
dts->samples = dts_samples (dts->state);
dts->bit_rate = -1;
dts->sample_rate = -1;
dts->stream_channels = 0;
/* FIXME force stereo for now */
dts->request_channels = DTS_CHANNEL;
dts->using_channels = 0;
dts->level = 1;
dts->bias = 0;
dts->current_ts = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
dts->samples = NULL;
if (dts->cache) {
gst_buffer_unref (dts->cache);
dts->cache = NULL;
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
dts_free (dts->state);
dts->state = NULL;
break;
default:
break;
}
return ret;
}
static void
gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
dts->dynamic_range_compression = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
g_value_set_boolean (value, dts->dynamic_range_compression);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
GST_TYPE_DTSDEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"dtsdec",
"Decodes DTS audio streams",
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);