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New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste...

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Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
This commit is contained in:
Ronald S. Bultje 2003-10-01 13:14:51 +00:00
parent 7fe85ee0b0
commit 0ea59b7a7b
20 changed files with 312 additions and 484 deletions

3
configure.ac
View file

@ -262,7 +262,7 @@ GST_PLUGINS_ALL="\
ac3parse adder audioscale auparse avi \ ac3parse adder audioscale auparse avi \
asfdemux audioconvert cdxaparse chart \ asfdemux audioconvert cdxaparse chart \
cutter debug deinterlace effectv festival \ cutter debug deinterlace effectv festival \
filter flx goom intfloat law level median mixmatrix \ filter flx goom id3 intfloat law level median mixmatrix \
mpeg1sys mpeg1videoparse mpeg2enc mpeg2sub \ mpeg1sys mpeg1videoparse mpeg2enc mpeg2sub \
mpegaudio mpegaudioparse mpegstream mpegtypes \ mpegaudio mpegaudioparse mpegstream mpegtypes \
monoscope oneton overlay passthrough playondemand qtdemux \ monoscope oneton overlay passthrough playondemand qtdemux \
@ -1180,6 +1180,7 @@ gst/festival/Makefile
gst/filter/Makefile gst/filter/Makefile
gst/flx/Makefile gst/flx/Makefile
gst/goom/Makefile gst/goom/Makefile
gst/id3/Makefile
gst/intfloat/Makefile gst/intfloat/Makefile
gst/law/Makefile gst/law/Makefile
gst/level/Makefile gst/level/Makefile

5
ext/Makefile.am
View file

@ -278,8 +278,9 @@ SUBDIRS=$(A52DEC_DIR) $(AALIB_DIR) $(ALSA_DIR) \
$(ARTS_DIR) $(ARTSC_DIR) $(AUDIOFILE_DIR) \ $(ARTS_DIR) $(ARTSC_DIR) $(AUDIOFILE_DIR) \
$(CDPARANOIA_DIR) $(DIVX_DIR) \ $(CDPARANOIA_DIR) $(DIVX_DIR) \
$(DVDREAD_DIR) $(DVDNAV_DIR) $(ESD_DIR) $(MAS_DIR) \ $(DVDREAD_DIR) $(DVDNAV_DIR) $(ESD_DIR) $(MAS_DIR) \
$(FFMPEG_DIR) $(FLAC_DIR) $(GDK_PIXBUF_DIR) $(GNOMEVFS_DIR) $(GSM_DIR) \ $(FFMPEG_DIR) $(FLAC_DIR) $(GDK_PIXBUF_DIR) \
$(HERMES_DIR) $(JACK_DIR) $(JPEG_DIR) \ $(GNOMEVFS_DIR) $(GSM_DIR) $(HERMES_DIR) \
$(JACK_DIR) $(JPEG_DIR) \
$(LADSPA_DIR) $(LAME_DIR) $(LCS_DIR) \ $(LADSPA_DIR) $(LAME_DIR) $(LCS_DIR) \
$(LIBDV_DIR) $(LIBFAME_DIR) $(LIBPNG_DIR) \ $(LIBDV_DIR) $(LIBFAME_DIR) $(LIBPNG_DIR) \
$(MAD_DIR) $(MATROSKA_DIR) $(MIKMOD_DIR) \ $(MAD_DIR) $(MATROSKA_DIR) $(MIKMOD_DIR) \

4
ext/a52dec/gsta52dec.c
View file

@ -619,10 +619,6 @@ plugin_init (GModule * module, GstPlugin * plugin)
{ {
GstElementFactory *factory; GstElementFactory *factory;
/* this filter needs the bytestream package */
if (!gst_library_load ("gstbytestream"))
return FALSE;
/* create an elementfactory for the a52dec element */ /* create an elementfactory for the a52dec element */
factory = gst_element_factory_new ("a52dec", GST_TYPE_A52DEC, &gst_a52dec_details); factory = gst_element_factory_new ("a52dec", GST_TYPE_A52DEC, &gst_a52dec_details);
g_return_val_if_fail (factory != NULL, FALSE); g_return_val_if_fail (factory != NULL, FALSE);

2
ext/a52dec/gsta52dec.h
View file

@ -23,7 +23,7 @@
#include <config.h> #include <config.h>
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/bytestream/bytestream.h> #include <gst/gstbytestream.h>
G_BEGIN_DECLS G_BEGIN_DECLS

7
ext/mad/gstmad.c
View file

@ -139,6 +139,7 @@ GST_PAD_TEMPLATE_FACTORY (mad_sink_template_factory,
"mad_sink", "mad_sink",
"audio/mpeg", "audio/mpeg",
/* we don't need channel/rate ... */ /* we don't need channel/rate ... */
"mpegversion", GST_PROPS_INT (1),
"layer", GST_PROPS_INT_RANGE (1, 3) "layer", GST_PROPS_INT_RANGE (1, 3)
) )
) )
@ -1184,15 +1185,13 @@ gst_mad_chain (GstPad *pad, GstBuffer *buffer)
mad->stream.bufend - mad->stream.this_frame); mad->stream.bufend - mad->stream.this_frame);
if (tagsize > mad->tempsize) { if (tagsize > mad->tempsize) {
GST_INFO ( GST_INFO ("mad: got partial id3 tag in buffer, skipping");
"mad: got partial id3 tag in buffer, skipping");
} }
else if (tagsize > 0) { else if (tagsize > 0) {
struct id3_tag *tag; struct id3_tag *tag;
id3_byte_t const *data; id3_byte_t const *data;
GST_INFO ( GST_INFO ("mad: got ID3 tag size %ld", tagsize);
"mad: got ID3 tag size %ld", tagsize);
data = mad->stream.this_frame; data = mad->stream.this_frame;

149
gst/asfdemux/gstasfdemux.c
View file

@ -35,123 +35,41 @@ static GstElementDetails gst_asf_demux_details = {
"(C) 2002", "(C) 2002",
}; };
static GstCaps* asf_asf_type_find (GstBuffer *buf, gpointer private); static GstCaps* asf_type_find (GstByteStream *bs, gpointer private);
static GstCaps* asf_wma_type_find (GstBuffer *buf, gpointer private);
static GstCaps* asf_wax_type_find (GstBuffer *buf, gpointer private);
static GstCaps* asf_wmv_type_find (GstBuffer *buf, gpointer private);
static GstCaps* asf_wvx_type_find (GstBuffer *buf, gpointer private);
static GstCaps* asf_wm_type_find (GstBuffer *buf, gpointer private);
/* typefactory for 'asf' */ /* typefactory for 'asf' */
static GstTypeDefinition asf_type_definitions[] = { static GstTypeDefinition asf_type_definition = {
{ "asfdemux_video/asf", "asfdemux_video/asf",
"video/x-ms-asf", "video/x-ms-asf",
".asf .asx", /* note: asx/wax/wmx are XML files, we don't handle them */
asf_asf_type_find }, ".asf .wma .wmv .wm",
{ "asfdemux_video/wma", asf_type_find,
"video/x-ms-wma",
".wma",
asf_wma_type_find },
{ "asfdemux_video/wax",
"video/x-ms-wax",
".wax",
asf_wax_type_find },
{ "asfdemux_video/wmv",
"video/x-ms-wmv",
".wmv",
asf_wmv_type_find },
{ "asfdemux_video/wvx",
"video/x-ms-wvx",
".wvx",
asf_wvx_type_find },
{ "asfdemux_video/wm",
"video/x-ms-wm",
".wm",
asf_wm_type_find },
{ NULL, NULL, NULL, NULL }
}; };
static GstCaps* static GstCaps*
asf_asf_type_find (GstBuffer *buf, gpointer private) asf_type_find (GstByteStream *bs, gpointer private)
{ {
GstCaps *new; GstCaps *new = NULL;
GstBuffer *buf = NULL;
new = gst_caps_new ( if (gst_bytestream_peek (bs, &buf, 16) == 16) {
"asf_type_find", guint32 uid1 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[0]),
"video/x-ms-asf", uid2 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[1]),
gst_props_new ("asfversion", uid3 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[2]),
GST_PROPS_INT (1), uid4 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[3]);
NULL));
return new;
}
static GstCaps* if (uid1 == 0x75B22630 && uid2 == 0x11CF668E &&
asf_wma_type_find (GstBuffer *buf, gpointer private) uid3 == 0xAA00D9A6 && uid4 == 0x6CCE6200) {
{ new = GST_CAPS_NEW ("asf_type_find",
GstCaps *new; "video/x-ms-asf",
NULL);
}
}
new = gst_caps_new ( if (buf != NULL) {
"asf_type_find", gst_buffer_unref (buf);
"video/x-ms-asf", }
gst_props_new ("asfversion",
GST_PROPS_INT (1),
NULL));
return new;
}
static GstCaps*
asf_wax_type_find (GstBuffer *buf, gpointer private)
{
GstCaps *new;
new = gst_caps_new (
"asf_type_find",
"video/x-ms-asf",
gst_props_new ("asfversion",
GST_PROPS_INT (1),
NULL));
return new;
}
static GstCaps*
asf_wmv_type_find (GstBuffer *buf, gpointer private)
{
GstCaps *new;
new = gst_caps_new (
"asf_type_find",
"video/x-ms-asf",
gst_props_new ("asfversion",
GST_PROPS_INT (1),
NULL));
return new;
}
static GstCaps*
asf_wvx_type_find (GstBuffer *buf, gpointer private)
{
GstCaps *new;
new = gst_caps_new (
"asf_type_find",
"video/x-ms-asf",
gst_props_new ("asfversion",
GST_PROPS_INT (1),
NULL));
return new;
}
static GstCaps*
asf_wm_type_find (GstBuffer *buf, gpointer private)
{
GstCaps *new;
new = gst_caps_new (
"asf_type_find",
"video/x-ms-asf",
gst_props_new ("asfversion",
GST_PROPS_INT (1),
NULL));
return new; return new;
} }
@ -1644,6 +1562,7 @@ static gboolean
plugin_init (GModule *module, GstPlugin *plugin) plugin_init (GModule *module, GstPlugin *plugin)
{ {
GstElementFactory *factory; GstElementFactory *factory;
GstTypeFactory *type;
gint i = 0; gint i = 0;
GstCaps *audcaps = NULL, *vidcaps = NULL, *temp; GstCaps *audcaps = NULL, *vidcaps = NULL, *temp;
guint32 vid_list[] = { guint32 vid_list[] = {
@ -1670,23 +1589,13 @@ plugin_init (GModule *module, GstPlugin *plugin)
-1 /* end */ -1 /* end */
}; };
/* this filter needs bytestream */
if (!gst_library_load ("gstbytestream")) {
GST_INFO ("asfdemux: could not load support library: 'gstbytestream'\n");
return FALSE;
}
/* create an elementfactory for the asf_demux element */ /* create an elementfactory for the asf_demux element */
factory = gst_element_factory_new ("asfdemux",GST_TYPE_ASF_DEMUX, factory = gst_element_factory_new ("asfdemux",GST_TYPE_ASF_DEMUX,
&gst_asf_demux_details); &gst_asf_demux_details);
while (asf_type_definitions[i].name) { /* type finding */
GstTypeFactory *type; type = gst_type_factory_new (&asf_type_definition);
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
type = gst_type_factory_new (&asf_type_definitions[i]);
//gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
i++;
}
g_return_val_if_fail (factory != NULL, FALSE); g_return_val_if_fail (factory != NULL, FALSE);
gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_NONE); gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_NONE);

2
gst/asfdemux/gstasfdemux.h
View file

@ -24,7 +24,7 @@
#include <config.h> #include <config.h>
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/riff/riff.h> #include <gst/riff/riff.h>
#include <gst/bytestream/bytestream.h> #include <gst/gstbytestream.h>
#include "asfheaders.h" #include "asfheaders.h"
G_BEGIN_DECLS G_BEGIN_DECLS

19
gst/asfdemux/gstasfmux.c
View file

@ -414,6 +414,7 @@ gst_asfmux_vidsinkconnect (GstPad *pad, GstCaps *vscaps)
return GST_PAD_LINK_REFUSED; return GST_PAD_LINK_REFUSED;
done: done:
stream->bitrate = 1024 * 1024;
stream->header.video.format.size = stream->header.video.stream.size; stream->header.video.format.size = stream->header.video.stream.size;
stream->header.video.format.width = stream->header.video.stream.width; stream->header.video.format.width = stream->header.video.stream.width;
stream->header.video.format.height = stream->header.video.stream.height; stream->header.video.format.height = stream->header.video.stream.height;
@ -465,7 +466,6 @@ gst_asfmux_audsinkconnect (GstPad *pad, GstCaps *vscaps)
stream->header.audio.sample_rate = rate; stream->header.audio.sample_rate = rate;
stream->header.audio.channels = channels; stream->header.audio.channels = channels;
stream->bitrate = 0; /* TODO */
if (!strcmp (mimetype, "audio/x-raw-int")) { if (!strcmp (mimetype, "audio/x-raw-int")) {
gint block, size; gint block, size;
@ -507,7 +507,7 @@ gst_asfmux_audsinkconnect (GstPad *pad, GstCaps *vscaps)
} }
stream->header.audio.block_align = 1; stream->header.audio.block_align = 1;
stream->header.audio.byte_rate = 0; stream->header.audio.byte_rate = 8 * 1024;
stream->header.audio.word_size = 16; stream->header.audio.word_size = 16;
stream->header.audio.size = 0; stream->header.audio.size = 0;
@ -521,6 +521,7 @@ gst_asfmux_audsinkconnect (GstPad *pad, GstCaps *vscaps)
return GST_PAD_LINK_REFUSED; return GST_PAD_LINK_REFUSED;
done: done:
stream->bitrate = stream->header.audio.byte_rate * 8;
return GST_PAD_LINK_OK; return GST_PAD_LINK_OK;
} }
@ -632,9 +633,11 @@ gst_asfmux_can_seek (GstAsfMux *asfmux)
const GstEventMask *masks = gst_pad_get_event_masks (GST_PAD_PEER (asfmux->srcpad)); const GstEventMask *masks = gst_pad_get_event_masks (GST_PAD_PEER (asfmux->srcpad));
/* this is for stream or file-storage */ /* this is for stream or file-storage */
while (masks->type != 0) { while (masks != NULL && masks->type != 0) {
if (masks->type == GST_EVENT_SEEK) { if (masks->type == GST_EVENT_SEEK) {
return TRUE; return TRUE;
} else {
masks++;
} }
} }
@ -724,7 +727,7 @@ gst_asfmux_put_buffer (GstBuffer *packet,
guint8 *data, guint8 *data,
guint length) guint length)
{ {
if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > length) { if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= length) {
guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet); guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
memcpy (pos, data, length); memcpy (pos, data, length);
GST_BUFFER_SIZE (packet) += length; GST_BUFFER_SIZE (packet) += length;
@ -737,7 +740,7 @@ static void
gst_asfmux_put_byte (GstBuffer *packet, gst_asfmux_put_byte (GstBuffer *packet,
guint8 data) guint8 data)
{ {
if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) { if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet); guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
* (guint8 *) pos = data; * (guint8 *) pos = data;
GST_BUFFER_SIZE (packet) += 1; GST_BUFFER_SIZE (packet) += 1;
@ -750,7 +753,7 @@ static void
gst_asfmux_put_le16 (GstBuffer *packet, gst_asfmux_put_le16 (GstBuffer *packet,
guint16 data) guint16 data)
{ {
if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) { if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet); guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
* (guint16 *) pos = GUINT16_TO_LE (data); * (guint16 *) pos = GUINT16_TO_LE (data);
GST_BUFFER_SIZE (packet) += 2; GST_BUFFER_SIZE (packet) += 2;
@ -763,7 +766,7 @@ static void
gst_asfmux_put_le32 (GstBuffer *packet, gst_asfmux_put_le32 (GstBuffer *packet,
guint32 data) guint32 data)
{ {
if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) { if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet); guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
* (guint32 *) pos = GUINT32_TO_LE (data); * (guint32 *) pos = GUINT32_TO_LE (data);
GST_BUFFER_SIZE (packet) += 4; GST_BUFFER_SIZE (packet) += 4;
@ -776,7 +779,7 @@ static void
gst_asfmux_put_le64 (GstBuffer *packet, gst_asfmux_put_le64 (GstBuffer *packet,
guint64 data) guint64 data)
{ {
if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) { if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet); guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
* (guint64 *) pos = GUINT64_TO_LE (data); * (guint64 *) pos = GUINT64_TO_LE (data);
GST_BUFFER_SIZE (packet) += 8; GST_BUFFER_SIZE (packet) += 8;

10
gst/mpegaudioparse/Makefile.am
View file

@ -1,17 +1,9 @@
#FIXME clean me up a bit plugin_LTLIBRARIES = libgstmpegaudioparse.la
plugin_LTLIBRARIES = libgstmpegaudioparse.la libgstmp3types.la
libgstmpegaudioparse_la_SOURCES = gstmpegaudioparse.c libgstmpegaudioparse_la_SOURCES = gstmpegaudioparse.c
libgstmpegaudioparse_la_CFLAGS = $(GST_CFLAGS) libgstmpegaudioparse_la_CFLAGS = $(GST_CFLAGS)
libgstmpegaudioparse_la_LIBADD = libgstmpegaudioparse_la_LIBADD =
libgstmpegaudioparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstmpegaudioparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstmp3types_la_SOURCES = gstmp3types.c
libgstmp3types_la_CFLAGS = $(GST_CFLAGS)
libgstmp3types_la_LIBADD =
libgstmp3types_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstmpegaudioparse.h noinst_HEADERS = gstmpegaudioparse.h
EXTRA_DIST = README EXTRA_DIST = README

251
gst/mpegaudioparse/gstmp3types.c
View file

@ -1,251 +0,0 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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.
*/
/*#define DEBUG_ENABLED */
#include <gst/gst.h>
#include <string.h> /* memcmp */
static GstCaps* mp3_type_find(GstBuffer *buf, gpointer private);
static GstCaps* mp3_type_find_stream(GstBuffer *buf, gpointer private);
static GstTypeDefinition mp3type_definitions[] = {
{ "mp3types_audio/mpeg", "audio/mpeg", ".mp3 .mp2 .mp1 .mpga", mp3_type_find },
{ "mp3types_stream_audio/mpeg", "audio/mpeg", ".mp3 .mp2 .mp1 .mpga", mp3_type_find_stream },
{ NULL, NULL, NULL, NULL },
};
static GstCaps*
mp3_type_find(GstBuffer *buf, gpointer private)
{
guint8 *data;
gint size, layer;
guint32 head;
GstCaps *caps;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG ("mp3typefind: typefind");
/* gracefully ripped from libid3 */
if (size >= 3 &&
data[0] == 'T' && data[1] == 'A' && data[2] == 'G') {
/* ID V1 tags */
data += 128;
size -= 128;
GST_DEBUG ("mp3typefind: detected ID3 Tag V1");
} else if (size >= 10 &&
(data[0] == 'I' && data[1] == 'D' && data[2] == '3') &&
data[3] < 0xff && data[4] < 0xff &&
data[6] < 0x80 && data[7] < 0x80 && data[8] < 0x80 && data[9] < 0x80)
{
guint32 skip = 0;
skip = (skip << 7) | (data[6] & 0x7f);
skip = (skip << 7) | (data[7] & 0x7f);
skip = (skip << 7) | (data[8] & 0x7f);
skip = (skip << 7) | (data[9] & 0x7f);
/* include size of header */
skip += 10;
/* footer present? (only available since version 4) */
if (data[3] > 3 && (data[5] & 0x10))
skip += 10;
GST_DEBUG ("mp3typefind: detected ID3 Tag V2 with %u bytes", skip);
size -= skip;
data += skip;
}
if (size < 4)
return NULL;
/* now with the right postion, do typefinding */
head = GUINT32_FROM_BE(*((guint32 *)data));
if ((head & 0xffe00000) != 0xffe00000)
return NULL;
if (!(layer = ((head >> 17) & 3)))
return NULL;
layer = 4 - layer;
if (((head >> 12) & 0xf) == 0xf)
return NULL;
if (!((head >> 12) & 0xf))
return NULL;
if (((head >> 10) & 0x3) == 0x3)
return NULL;
caps = GST_CAPS_NEW ("mp3_type_find", "audio/mpeg", "layer", GST_PROPS_INT (layer));
return caps;
}
static guint mp3types_bitrates[2][3][16] =
{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, },
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } },
{ {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } },
};
static guint mp3types_freqs[3][3] =
{ {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}};
static inline guint
mp3_type_frame_length_from_header (guint32 header, guint *put_layer)
{
guint length;
gulong samplerate, bitrate, layer, version;
/* we don't need extension, mode, copyright, original or emphasis for the frame length */
header >>= 9;
/* padding */
length = header & 0x1;
header >>= 1;
/* sampling frequency */
samplerate = header & 0x3;
if (samplerate == 3)
return 0;
header >>= 2;
/* bitrate index */
bitrate = header & 0xF;
if (bitrate == 15 || bitrate == 0)
return 0;
/* ignore error correction, too */
header >>= 5;
/* layer */
layer = 4 - (header & 0x3);
if (layer == 4)
return 0;
header >>= 2;
/* version */
version = header & 0x3;
if (version == 1)
return 0;
/* lookup */
bitrate = mp3types_bitrates[version == 3 ? 0 : 1][layer - 1][bitrate];
samplerate = mp3types_freqs[version > 0 ? version - 1 : 0][samplerate];
/* calculating */
if (layer == 1) {
length = ((12000 * bitrate / samplerate) + length) * 4;
} else {
length += ((layer == 3 && version == 0) ? 144000 : 72000) * bitrate / samplerate;
}
GST_DEBUG ("Calculated mad frame length of %u bytes", length);
GST_DEBUG ("samplerate = %lu - bitrate = %lu - layer = %lu - version = %lu", samplerate, bitrate, layer, version);
if (put_layer)
*put_layer = layer;
return length;
}
/* increase this value when this function finds too many false positives */
/**
* The chance that random data is identified as a valid mp3 header is 63 / 2^18
* (0.024%) per try. This makes the function for calculating false positives
* 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize)
* This has the following probabilities of false positives:
* bufsize MIN_HEADERS
* (bytes) 1 2 3 4
* 4096 62.6% 0.02% 0% 0%
* 16384 98% 0.09% 0% 0%
* 1 MiB 100% 5.88% 0% 0%
* 1 GiB 100% 100% 1.44% 0%
* 1 TiB 100% 100% 100% 0.35%
* This means that the current choice (3 headers by most of the time 4096 byte
* buffers is pretty safe for now.
* It is however important to note that in a worst case example a buffer of size
* 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3
* bytes is needed to reliable find the mp3 stream in a buffer when scanning
* starts at a random position. This is currently (4323 bytes) slightly above
* the default buffer size. But you rarely hit the worst case - average mp3
* frames are in the 500 bytes range.
*/
#define GST_MP3_TYPEFIND_MIN_HEADERS 3
static GstCaps*
mp3_type_find_stream (GstBuffer *buf, gpointer private)
{
guint8 *data;
guint size;
guint32 head;
gint layer = 0;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
while (size >= 4) {
head = GUINT32_FROM_BE(*((guint32 *)data));
if ((head & 0xffe00000) == 0xffe00000) {
guint length;
guint prev_layer = 0;
guint found = 0; /* number of valid headers found */
guint pos = 0;
do {
if ((length = mp3_type_frame_length_from_header (head, &layer))) {
if (prev_layer && prev_layer != layer)
break;
prev_layer = layer;
pos += length;
found++;
if (pos + 4 >= size) {
if (found >= GST_MP3_TYPEFIND_MIN_HEADERS)
goto success;
}
head = GUINT32_FROM_BE(*((guint32 *) &(data[pos])));
if ((head & 0xffe00000) != 0xffe00000)
break;
} else {
break;
}
} while (TRUE);
}
data++;
size--;
}
return NULL;
success:
g_assert (layer);
return GST_CAPS_NEW ("mp3_type_find", "audio/mpeg", "layer", GST_PROPS_INT (layer));
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
gint i=0;
while (mp3type_definitions[i].name) {
GstTypeFactory *type;
type = gst_type_factory_new (&mp3type_definitions[i]);
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
i++;
}
/* gst_info("gsttypes: loaded %d mp3 types\n",i); */
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"mp3types",
plugin_init
};

288
gst/mpegaudioparse/gstmpegaudioparse.c
View file

@ -35,6 +35,15 @@ static GstElementDetails mp3parse_details = {
"(C) 1999", "(C) 1999",
}; };
static GstCaps * mp3_type_find (GstByteStream *bs, gpointer data);
static GstTypeDefinition mp3type_definition = {
"mp3_audio/mpeg",
"audio/mpeg",
".mp3 .mp2 .mp1 .mpga",
mp3_type_find,
};
static GstPadTemplate* static GstPadTemplate*
mp3_src_factory (void) mp3_src_factory (void)
{ {
@ -115,11 +124,221 @@ gst_mp3parse_get_type(void) {
0, 0,
(GInstanceInitFunc)gst_mp3parse_init, (GInstanceInitFunc)gst_mp3parse_init,
}; };
mp3parse_type = g_type_register_static(GST_TYPE_ELEMENT, "GstMPEGAudioParse", &mp3parse_info, 0); mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstMPEGAudioParse",
&mp3parse_info, 0);
} }
return mp3parse_type; return mp3parse_type;
} }
static guint mp3types_bitrates[2][3][16] =
{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, },
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } },
{ {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } },
};
static guint mp3types_freqs[3][3] =
{ {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}};
static inline guint
mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
guint *put_channels, guint *put_bitrate,
guint *put_samplerate)
{
guint length;
gulong mode, samplerate, bitrate, layer, version, channels;
/* we don't need extension, copyright, original or
* emphasis for the frame length */
header >>= 6;
/* mode */
mode = header & 0x3;
header >>= 3;
/* padding */
length = header & 0x1;
header >>= 1;
/* sampling frequency */
samplerate = header & 0x3;
if (samplerate == 3)
return 0;
header >>= 2;
/* bitrate index */
bitrate = header & 0xF;
if (bitrate == 15 || bitrate == 0)
return 0;
/* ignore error correction, too */
header >>= 5;
/* layer */
layer = 4 - (header & 0x3);
if (layer == 4)
return 0;
header >>= 2;
/* version */
version = header & 0x3;
if (version == 1)
return 0;
/* lookup */
channels = (mode == 3) ? 1 : 2;
bitrate = mp3types_bitrates[version == 3 ? 0 : 1][layer - 1][bitrate];
samplerate = mp3types_freqs[version > 0 ? version - 1 : 0][samplerate];
/* calculating */
if (layer == 1) {
length = ((12000 * bitrate / samplerate) + length) * 4;
} else {
length += ((layer == 3 && version == 0) ? 144000 : 72000) * bitrate / samplerate;
}
GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
GST_DEBUG ("samplerate = %lu - bitrate = %lu - layer = %lu - version = %lu"
" - channels = %lu",
samplerate, bitrate, layer, version, channels);
if (put_layer)
*put_layer = layer;
if (put_channels)
*put_channels = channels;
if (put_bitrate)
*put_bitrate = bitrate;
if (put_samplerate)
*put_samplerate = samplerate;
return length;
}
/**
* The chance that random data is identified as a valid mp3 header is 63 / 2^18
* (0.024%) per try. This makes the function for calculating false positives
* 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize)
* This has the following probabilities of false positives:
* bufsize MIN_HEADERS
* (bytes) 1 2 3 4
* 4096 62.6% 0.02% 0% 0%
* 16384 98% 0.09% 0% 0%
* 1 MiB 100% 5.88% 0% 0%
* 1 GiB 100% 100% 1.44% 0%
* 1 TiB 100% 100% 100% 0.35%
* This means that the current choice (3 headers by most of the time 4096 byte
* buffers is pretty safe for now.
*
* The max. size of each frame is 1440 bytes, which means that for N frames
* to be detected, we need 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3 of data.
* Assuming we step into the stream right after the frame header, this
* means we need 1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3 bytes
* of data (5762) to always detect any mp3.
*/
/* increase this value when this function finds too many false positives */
#define GST_MP3_TYPEFIND_MIN_HEADERS 3
#define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3)
static GstCaps *
mp3_caps_create (guint layer, guint channels,
guint bitrate, guint samplerate)
{
GstCaps *new;
g_assert (layer);
g_assert (samplerate);
g_assert (bitrate);
g_assert (channels);
new = GST_CAPS_NEW ("mp3_type_find",
"audio/mpeg",
"mpegversion", GST_PROPS_INT (1),
"layer", GST_PROPS_INT (layer),
/*"bitrate", GST_PROPS_INT (bitrate),*/
"rate", GST_PROPS_INT (samplerate),
"channels", GST_PROPS_INT (channels));
return new;
}
static GstCaps *
mp3_type_find (GstByteStream *bs, gpointer private)
{
GstBuffer *buf = NULL;
GstCaps *new = NULL;
guint8 *data;
guint size;
guint32 head;
guint layer = 0, bitrate = 0, samplerate = 0, channels = 0;
/* note that even if we don't get the requested size,
* it might still be a (very small) mp3 */
gst_bytestream_peek (bs, &buf, GST_MP3_TYPEFIND_MIN_DATA);
if (!buf) {
goto done;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
while (size >= 4) {
head = GUINT32_FROM_BE(*((guint32 *)data));
if ((head & 0xffe00000) == 0xffe00000) {
guint length;
guint prev_layer = 0, prev_bitrate = 0,
prev_channels = 0, prev_samplerate = 0;
guint found = 0; /* number of valid headers found */
guint pos = 0;
do {
if (!(length = mp3_type_frame_length_from_header (head, &layer,
&channels, &bitrate,
&samplerate))) {
break;
}
if ((prev_layer && prev_layer != layer) || !layer ||
(prev_bitrate && prev_bitrate != bitrate) || !bitrate ||
(prev_samplerate && prev_samplerate != samplerate) || !samplerate ||
(prev_channels && prev_channels != channels) || !channels) {
/* this means an invalid property, or a change, which likely
* indicates that this is not a mp3 but just a random bytestream */
break;
}
prev_layer = layer;
prev_bitrate = bitrate;
prev_channels = channels;
prev_samplerate = samplerate;
pos += length;
if (++found >= GST_MP3_TYPEFIND_MIN_HEADERS) {
/* we're pretty sure that this is mp3 now */
new = mp3_caps_create (layer, channels, bitrate, samplerate);
goto done;
}
/* and now, find a new head */
head = GUINT32_FROM_BE(*((guint32 *) &(data[pos])));
if ((head & 0xffe00000) != 0xffe00000)
break;
} while (TRUE);
}
data++;
size--;
}
done:
if (buf != NULL) {
gst_buffer_unref (buf);
}
return new;
}
static void static void
gst_mp3parse_class_init (GstMPEGAudioParseClass *klass) gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
{ {
@ -247,7 +466,10 @@ gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf))); header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf)));
GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf ); GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf );
#define HDRMASK ~( (0xF<<12)/*bitrate*/ | (1<<9)/*padding*/ | (3<<4)/*mode extension*/ ) /* mask the bits which are allowed to differ between frames */ /* mask the bits which are allowed to differ between frames */
#define HDRMASK ~((0xF << 12) /* bitrate */ | \
(0x1 << 9) /* padding */ | \
(0x3 << 4)) /*mode extension*/
if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */ if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */
GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf ); GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf );
@ -309,56 +531,23 @@ gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
} }
} }
static int mp3parse_tabsel[2][3][16] =
{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, },
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } },
{ {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } },
};
static long mp3parse_freqs[9] =
{44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000};
static long static long
bpf_from_header (GstMPEGAudioParse *parse, unsigned long header) bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
{ {
int layer_index,layer,lsf,samplerate_index,padding,mode; guint bitrate, layer, rate, channels, length;
long bpf;
gint channels, rate;
/*mpegver = (header >> 19) & 0x3; // don't need this for bpf */ if (!(length = mp3_type_frame_length_from_header (header, &layer,
layer_index = (header >> 17) & 0x3; &channels,
layer = 4 - layer_index; &bitrate, &rate))) {
lsf = (header & (1 << 20)) ? ((header & (1 << 19)) ? 0 : 1) : 1; return 0;
parse->bit_rate = mp3parse_tabsel[lsf][layer - 1][((header >> 12) & 0xf)];
samplerate_index = (header >> 10) & 0x3;
padding = (header >> 9) & 0x1;
mode = (header >> 6) & 0x3;
if (layer == 1) {
bpf = parse->bit_rate * 12000;
bpf /= mp3parse_freqs[samplerate_index];
bpf = ((bpf + padding) << 2);
} else {
bpf = parse->bit_rate * 144000;
bpf /= mp3parse_freqs[samplerate_index];
bpf += padding;
} }
channels = (mode == 3) ? 1 : 2;
rate = mp3parse_freqs[samplerate_index];
if (channels != parse->channels || if (channels != parse->channels ||
rate != parse->rate || rate != parse->rate ||
layer != parse->layer) { layer != parse->layer ||
GstCaps *caps = GST_CAPS_NEW ("mp3parse_src", bitrate != parse->bit_rate) {
"audio/mpeg", GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
"mpegversion", GST_PROPS_INT (1),
"layer", GST_PROPS_INT (layer),
"channels", GST_PROPS_INT (channels),
"rate", GST_PROPS_INT (rate));
if (gst_pad_try_set_caps(parse->srcpad, caps) <= 0) { if (gst_pad_try_set_caps(parse->srcpad, caps) <= 0) {
gst_element_error (GST_ELEMENT (parse), gst_element_error (GST_ELEMENT (parse),
"mp3parse: failed to negotiate format with next element"); "mp3parse: failed to negotiate format with next element");
@ -367,12 +556,10 @@ bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
parse->channels = channels; parse->channels = channels;
parse->layer = layer; parse->layer = layer;
parse->rate = rate; parse->rate = rate;
parse->bit_rate = bitrate;
} }
/*g_print("%08x: layer %d lsf %d bitrate %d samplerate_index %d padding %d - bpf %d\n", */ return length;
/*header,layer,lsf,bitrate,samplerate_index,padding,bpf); */
return bpf;
} }
static gboolean static gboolean
@ -470,6 +657,7 @@ static gboolean
plugin_init (GModule *module, GstPlugin *plugin) plugin_init (GModule *module, GstPlugin *plugin)
{ {
GstElementFactory *factory; GstElementFactory *factory;
GstTypeFactory *type;
/* create an elementfactory for the mp3parse element */ /* create an elementfactory for the mp3parse element */
factory = gst_element_factory_new ("mp3parse", factory = gst_element_factory_new ("mp3parse",
@ -485,6 +673,10 @@ plugin_init (GModule *module, GstPlugin *plugin)
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
/* type finding */
type = gst_type_factory_new (&mp3type_definition);
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
return TRUE; return TRUE;
} }

4
gst/mpegstream/gstmpegdemux.c
View file

@ -1235,10 +1235,6 @@ gst_mpeg_demux_plugin_init (GModule *module, GstPlugin *plugin)
{ {
GstElementFactory *factory; GstElementFactory *factory;
/* this filter needs the bytestream package */
if (!gst_library_load ("gstbytestream"))
return FALSE;
/* create an elementfactory for the mpeg_demux element */ /* create an elementfactory for the mpeg_demux element */
factory = gst_element_factory_new ("mpegdemux", GST_TYPE_MPEG_DEMUX, factory = gst_element_factory_new ("mpegdemux", GST_TYPE_MPEG_DEMUX,
&mpeg_demux_details); &mpeg_demux_details);

2
gst/mpegstream/gstmpegpacketize.h
View file

@ -24,7 +24,7 @@
#include <config.h> #include <config.h>
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/bytestream/bytestream.h> #include <gst/gstbytestream.h>
#ifdef __cplusplus #ifdef __cplusplus

4
gst/mpegstream/gstmpegparse.c
View file

@ -915,10 +915,6 @@ gst_mpeg_parse_plugin_init (GModule *module, GstPlugin *plugin)
{ {
GstElementFactory *factory; GstElementFactory *factory;
/* this filter needs the bytestream package */
if (!gst_library_load ("gstbytestream"))
return FALSE;
/* create an elementfactory for the mpeg_parse element */ /* create an elementfactory for the mpeg_parse element */
factory = gst_element_factory_new ("mpegparse", GST_TYPE_MPEG_PARSE, factory = gst_element_factory_new ("mpegparse", GST_TYPE_MPEG_PARSE,
&mpeg_parse_details); &mpeg_parse_details);

2
gst/mpegstream/gstmpegparse.h
View file

@ -22,7 +22,7 @@
#define __MPEG_PARSE_H__ #define __MPEG_PARSE_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/bytestream/bytestream.h> #include <gst/gstbytestream.h>
#include "gstmpegpacketize.h" #include "gstmpegpacketize.h"
#ifdef __cplusplus #ifdef __cplusplus

4
gst/mpegstream/gstmpegstream.c
View file

@ -25,10 +25,6 @@
static gboolean static gboolean
plugin_init (GModule *module, GstPlugin *plugin) plugin_init (GModule *module, GstPlugin *plugin)
{ {
/* mpegdemux needs the bytestream package */
if (!gst_library_load ("gstbytestream"))
return FALSE;
/* short-circuit here; this is potentially dangerous since if the second /* short-circuit here; this is potentially dangerous since if the second
* or third init fails then the whole plug-in will be placed on the register * or third init fails then the whole plug-in will be placed on the register
* stack again and the first _init will be called more than once * stack again and the first _init will be called more than once

4
gst/mpegstream/gstrfc2250enc.c
View file

@ -328,10 +328,6 @@ gst_rfc2250_enc_plugin_init (GModule *module, GstPlugin *plugin)
{ {
GstElementFactory *factory; GstElementFactory *factory;
/* this filter needs the bytestream package */
if (!gst_library_load("gstbytestream"))
return FALSE;
/* create an elementfactory for the rfc2250_enc element */ /* create an elementfactory for the rfc2250_enc element */
factory = gst_element_factory_new ("rfc2250enc", GST_TYPE_RFC2250_ENC, factory = gst_element_factory_new ("rfc2250enc", GST_TYPE_RFC2250_ENC,
&rfc2250_enc_details); &rfc2250_enc_details);

2
gst/mpegstream/gstrfc2250enc.h
View file

@ -24,7 +24,7 @@
#include <config.h> #include <config.h>
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/bytestream/bytestream.h> #include <gst/gstbytestream.h>
#include "gstmpegpacketize.h" #include "gstmpegpacketize.h"

32
gst/realmedia/rmdemux.c
View file

@ -87,7 +87,7 @@ gst_rmdemux_details =
"(C) 2003", "(C) 2003",
}; };
static GstCaps* realmedia_type_find (GstBuffer *buf, gpointer private); static GstCaps* realmedia_type_find (GstByteStream *bs, gpointer private);
static GstTypeDefinition realmediadefinition = { static GstTypeDefinition realmediadefinition = {
"rmdemux_video/realmedia", "rmdemux_video/realmedia",
@ -195,21 +195,26 @@ gst_rmdemux_init (GstRMDemux *rmdemux)
} }
static GstCaps* static GstCaps*
realmedia_type_find (GstBuffer *buf, gpointer private) realmedia_type_find (GstByteStream *bs, gpointer private)
{ {
gchar *data = GST_BUFFER_DATA (buf); GstBuffer *buf = NULL;
GstCaps *new = NULL;
g_return_val_if_fail (data != NULL, NULL); if (gst_bytestream_peek (bs, &buf, 4) == 4) {
gchar *data = GST_BUFFER_DATA (buf);
if(GST_BUFFER_SIZE(buf) < 4){
return NULL; if (!strncmp (data, ".RMF", 4)) {
new = GST_CAPS_NEW ("realmedia_type_find",
"application/vnd.rn-realmedia",
NULL);
}
} }
if (strncmp (data, ".RMF", 4)==0) {
return gst_caps_new ("realmedia_type_find", if (buf != NULL) {
"application/vnd.rn-realmedia", gst_buffer_unref (buf);
NULL);
} }
return NULL;
return new;
} }
static gboolean static gboolean
@ -218,9 +223,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
GstElementFactory *factory; GstElementFactory *factory;
GstTypeFactory *type; GstTypeFactory *type;
if (!gst_library_load ("gstbytestream"))
return FALSE;
factory = gst_element_factory_new ("rmdemux", GST_TYPE_RMDEMUX, factory = gst_element_factory_new ("rmdemux", GST_TYPE_RMDEMUX,
&gst_rmdemux_details); &gst_rmdemux_details);
g_return_val_if_fail(factory != NULL, FALSE); g_return_val_if_fail(factory != NULL, FALSE);

2
gst/realmedia/rmdemux.h
View file

@ -22,7 +22,7 @@
#define __GST_RMDEMUX_H__ #define __GST_RMDEMUX_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/bytestream/bytestream.h> #include <gst/gstbytestream.h>
#ifdef __cplusplus #ifdef __cplusplus
extern "C" { extern "C" {