mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-03-28 03:45:39 +00:00
New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste...
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
parent
7fe85ee0b0
commit
0ea59b7a7b
20 changed files with 312 additions and 484 deletions
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@ -262,7 +262,7 @@ GST_PLUGINS_ALL="\
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ac3parse adder audioscale auparse avi \
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asfdemux audioconvert cdxaparse chart \
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cutter debug deinterlace effectv festival \
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filter flx goom intfloat law level median mixmatrix \
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filter flx goom id3 intfloat law level median mixmatrix \
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mpeg1sys mpeg1videoparse mpeg2enc mpeg2sub \
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mpegaudio mpegaudioparse mpegstream mpegtypes \
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monoscope oneton overlay passthrough playondemand qtdemux \
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@ -1180,6 +1180,7 @@ gst/festival/Makefile
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gst/filter/Makefile
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gst/flx/Makefile
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gst/goom/Makefile
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gst/id3/Makefile
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gst/intfloat/Makefile
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gst/law/Makefile
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gst/level/Makefile
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@ -278,8 +278,9 @@ SUBDIRS=$(A52DEC_DIR) $(AALIB_DIR) $(ALSA_DIR) \
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$(ARTS_DIR) $(ARTSC_DIR) $(AUDIOFILE_DIR) \
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$(CDPARANOIA_DIR) $(DIVX_DIR) \
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$(DVDREAD_DIR) $(DVDNAV_DIR) $(ESD_DIR) $(MAS_DIR) \
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$(FFMPEG_DIR) $(FLAC_DIR) $(GDK_PIXBUF_DIR) $(GNOMEVFS_DIR) $(GSM_DIR) \
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$(HERMES_DIR) $(JACK_DIR) $(JPEG_DIR) \
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$(FFMPEG_DIR) $(FLAC_DIR) $(GDK_PIXBUF_DIR) \
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$(GNOMEVFS_DIR) $(GSM_DIR) $(HERMES_DIR) \
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$(JACK_DIR) $(JPEG_DIR) \
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$(LADSPA_DIR) $(LAME_DIR) $(LCS_DIR) \
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$(LIBDV_DIR) $(LIBFAME_DIR) $(LIBPNG_DIR) \
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$(MAD_DIR) $(MATROSKA_DIR) $(MIKMOD_DIR) \
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@ -619,10 +619,6 @@ plugin_init (GModule * module, GstPlugin * plugin)
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{
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GstElementFactory *factory;
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/* this filter needs the bytestream package */
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if (!gst_library_load ("gstbytestream"))
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return FALSE;
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/* create an elementfactory for the a52dec element */
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factory = gst_element_factory_new ("a52dec", GST_TYPE_A52DEC, &gst_a52dec_details);
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g_return_val_if_fail (factory != NULL, FALSE);
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@ -23,7 +23,7 @@
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#include <config.h>
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#include <gst/gst.h>
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#include <gst/bytestream/bytestream.h>
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#include <gst/gstbytestream.h>
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G_BEGIN_DECLS
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@ -139,6 +139,7 @@ GST_PAD_TEMPLATE_FACTORY (mad_sink_template_factory,
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"mad_sink",
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"audio/mpeg",
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/* we don't need channel/rate ... */
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"mpegversion", GST_PROPS_INT (1),
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"layer", GST_PROPS_INT_RANGE (1, 3)
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)
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)
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@ -1184,15 +1185,13 @@ gst_mad_chain (GstPad *pad, GstBuffer *buffer)
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mad->stream.bufend - mad->stream.this_frame);
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if (tagsize > mad->tempsize) {
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GST_INFO (
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"mad: got partial id3 tag in buffer, skipping");
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GST_INFO ("mad: got partial id3 tag in buffer, skipping");
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}
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else if (tagsize > 0) {
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struct id3_tag *tag;
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id3_byte_t const *data;
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GST_INFO (
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"mad: got ID3 tag size %ld", tagsize);
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GST_INFO ("mad: got ID3 tag size %ld", tagsize);
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data = mad->stream.this_frame;
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@ -35,123 +35,41 @@ static GstElementDetails gst_asf_demux_details = {
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"(C) 2002",
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};
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static GstCaps* asf_asf_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* asf_wma_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* asf_wax_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* asf_wmv_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* asf_wvx_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* asf_wm_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* asf_type_find (GstByteStream *bs, gpointer private);
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/* typefactory for 'asf' */
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static GstTypeDefinition asf_type_definitions[] = {
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{ "asfdemux_video/asf",
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"video/x-ms-asf",
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".asf .asx",
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asf_asf_type_find },
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{ "asfdemux_video/wma",
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"video/x-ms-wma",
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".wma",
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asf_wma_type_find },
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{ "asfdemux_video/wax",
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"video/x-ms-wax",
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".wax",
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asf_wax_type_find },
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{ "asfdemux_video/wmv",
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"video/x-ms-wmv",
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".wmv",
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asf_wmv_type_find },
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{ "asfdemux_video/wvx",
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"video/x-ms-wvx",
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".wvx",
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asf_wvx_type_find },
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{ "asfdemux_video/wm",
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"video/x-ms-wm",
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".wm",
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asf_wm_type_find },
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{ NULL, NULL, NULL, NULL }
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static GstTypeDefinition asf_type_definition = {
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"asfdemux_video/asf",
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"video/x-ms-asf",
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/* note: asx/wax/wmx are XML files, we don't handle them */
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".asf .wma .wmv .wm",
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asf_type_find,
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};
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static GstCaps*
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asf_asf_type_find (GstBuffer *buf, gpointer private)
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asf_type_find (GstByteStream *bs, gpointer private)
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{
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GstCaps *new;
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GstCaps *new = NULL;
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GstBuffer *buf = NULL;
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new = gst_caps_new (
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"asf_type_find",
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"video/x-ms-asf",
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gst_props_new ("asfversion",
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GST_PROPS_INT (1),
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NULL));
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return new;
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}
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if (gst_bytestream_peek (bs, &buf, 16) == 16) {
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guint32 uid1 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[0]),
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uid2 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[1]),
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uid3 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[2]),
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uid4 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[3]);
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static GstCaps*
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asf_wma_type_find (GstBuffer *buf, gpointer private)
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{
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GstCaps *new;
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if (uid1 == 0x75B22630 && uid2 == 0x11CF668E &&
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uid3 == 0xAA00D9A6 && uid4 == 0x6CCE6200) {
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new = GST_CAPS_NEW ("asf_type_find",
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"video/x-ms-asf",
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NULL);
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}
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}
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new = gst_caps_new (
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"asf_type_find",
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"video/x-ms-asf",
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gst_props_new ("asfversion",
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GST_PROPS_INT (1),
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NULL));
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return new;
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}
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if (buf != NULL) {
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gst_buffer_unref (buf);
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}
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static GstCaps*
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asf_wax_type_find (GstBuffer *buf, gpointer private)
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{
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GstCaps *new;
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new = gst_caps_new (
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"asf_type_find",
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"video/x-ms-asf",
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gst_props_new ("asfversion",
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GST_PROPS_INT (1),
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NULL));
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return new;
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}
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static GstCaps*
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asf_wmv_type_find (GstBuffer *buf, gpointer private)
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{
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GstCaps *new;
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new = gst_caps_new (
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"asf_type_find",
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"video/x-ms-asf",
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gst_props_new ("asfversion",
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GST_PROPS_INT (1),
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NULL));
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return new;
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}
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static GstCaps*
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asf_wvx_type_find (GstBuffer *buf, gpointer private)
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{
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GstCaps *new;
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new = gst_caps_new (
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"asf_type_find",
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"video/x-ms-asf",
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gst_props_new ("asfversion",
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GST_PROPS_INT (1),
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NULL));
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return new;
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}
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static GstCaps*
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asf_wm_type_find (GstBuffer *buf, gpointer private)
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{
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GstCaps *new;
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new = gst_caps_new (
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"asf_type_find",
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"video/x-ms-asf",
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gst_props_new ("asfversion",
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GST_PROPS_INT (1),
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NULL));
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return new;
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}
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@ -1644,6 +1562,7 @@ static gboolean
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plugin_init (GModule *module, GstPlugin *plugin)
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{
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GstElementFactory *factory;
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GstTypeFactory *type;
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gint i = 0;
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GstCaps *audcaps = NULL, *vidcaps = NULL, *temp;
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guint32 vid_list[] = {
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@ -1670,23 +1589,13 @@ plugin_init (GModule *module, GstPlugin *plugin)
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-1 /* end */
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};
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/* this filter needs bytestream */
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if (!gst_library_load ("gstbytestream")) {
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GST_INFO ("asfdemux: could not load support library: 'gstbytestream'\n");
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return FALSE;
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}
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/* create an elementfactory for the asf_demux element */
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factory = gst_element_factory_new ("asfdemux",GST_TYPE_ASF_DEMUX,
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&gst_asf_demux_details);
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while (asf_type_definitions[i].name) {
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GstTypeFactory *type;
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type = gst_type_factory_new (&asf_type_definitions[i]);
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//gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
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i++;
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}
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/* type finding */
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type = gst_type_factory_new (&asf_type_definition);
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gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
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g_return_val_if_fail (factory != NULL, FALSE);
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gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_NONE);
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@ -24,7 +24,7 @@
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#include <config.h>
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#include <gst/gst.h>
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#include <gst/riff/riff.h>
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#include <gst/bytestream/bytestream.h>
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#include <gst/gstbytestream.h>
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#include "asfheaders.h"
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G_BEGIN_DECLS
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@ -414,6 +414,7 @@ gst_asfmux_vidsinkconnect (GstPad *pad, GstCaps *vscaps)
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return GST_PAD_LINK_REFUSED;
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done:
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stream->bitrate = 1024 * 1024;
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stream->header.video.format.size = stream->header.video.stream.size;
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stream->header.video.format.width = stream->header.video.stream.width;
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stream->header.video.format.height = stream->header.video.stream.height;
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@ -465,7 +466,6 @@ gst_asfmux_audsinkconnect (GstPad *pad, GstCaps *vscaps)
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stream->header.audio.sample_rate = rate;
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stream->header.audio.channels = channels;
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stream->bitrate = 0; /* TODO */
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if (!strcmp (mimetype, "audio/x-raw-int")) {
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gint block, size;
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@ -507,7 +507,7 @@ gst_asfmux_audsinkconnect (GstPad *pad, GstCaps *vscaps)
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}
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stream->header.audio.block_align = 1;
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stream->header.audio.byte_rate = 0;
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stream->header.audio.byte_rate = 8 * 1024;
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stream->header.audio.word_size = 16;
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stream->header.audio.size = 0;
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@ -521,6 +521,7 @@ gst_asfmux_audsinkconnect (GstPad *pad, GstCaps *vscaps)
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return GST_PAD_LINK_REFUSED;
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done:
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stream->bitrate = stream->header.audio.byte_rate * 8;
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return GST_PAD_LINK_OK;
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}
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@ -632,9 +633,11 @@ gst_asfmux_can_seek (GstAsfMux *asfmux)
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const GstEventMask *masks = gst_pad_get_event_masks (GST_PAD_PEER (asfmux->srcpad));
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/* this is for stream or file-storage */
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while (masks->type != 0) {
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while (masks != NULL && masks->type != 0) {
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if (masks->type == GST_EVENT_SEEK) {
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return TRUE;
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} else {
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masks++;
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}
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}
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@ -724,7 +727,7 @@ gst_asfmux_put_buffer (GstBuffer *packet,
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guint8 *data,
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guint length)
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{
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > length) {
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= length) {
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guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
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memcpy (pos, data, length);
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GST_BUFFER_SIZE (packet) += length;
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@ -737,7 +740,7 @@ static void
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gst_asfmux_put_byte (GstBuffer *packet,
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guint8 data)
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{
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) {
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
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guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
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* (guint8 *) pos = data;
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GST_BUFFER_SIZE (packet) += 1;
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@ -750,7 +753,7 @@ static void
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gst_asfmux_put_le16 (GstBuffer *packet,
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guint16 data)
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{
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) {
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
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guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
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* (guint16 *) pos = GUINT16_TO_LE (data);
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GST_BUFFER_SIZE (packet) += 2;
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@ -763,7 +766,7 @@ static void
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gst_asfmux_put_le32 (GstBuffer *packet,
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guint32 data)
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{
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) {
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
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guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
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* (guint32 *) pos = GUINT32_TO_LE (data);
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GST_BUFFER_SIZE (packet) += 4;
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@ -776,7 +779,7 @@ static void
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gst_asfmux_put_le64 (GstBuffer *packet,
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guint64 data)
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{
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) > sizeof (data)) {
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if ((GST_BUFFER_MAXSIZE (packet) - GST_BUFFER_SIZE (packet)) >= sizeof (data)) {
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guint8 *pos = GST_BUFFER_DATA (packet) + GST_BUFFER_SIZE (packet);
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* (guint64 *) pos = GUINT64_TO_LE (data);
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GST_BUFFER_SIZE (packet) += 8;
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@ -1,17 +1,9 @@
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#FIXME clean me up a bit
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plugin_LTLIBRARIES = libgstmpegaudioparse.la libgstmp3types.la
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plugin_LTLIBRARIES = libgstmpegaudioparse.la
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libgstmpegaudioparse_la_SOURCES = gstmpegaudioparse.c
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libgstmpegaudioparse_la_CFLAGS = $(GST_CFLAGS)
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libgstmpegaudioparse_la_LIBADD =
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libgstmpegaudioparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstmp3types_la_SOURCES = gstmp3types.c
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libgstmp3types_la_CFLAGS = $(GST_CFLAGS)
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libgstmp3types_la_LIBADD =
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libgstmp3types_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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noinst_HEADERS = gstmpegaudioparse.h
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EXTRA_DIST = README
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@ -1,251 +0,0 @@
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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||||
* 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
|
||||
};
|
||||
|
|
@ -35,6 +35,15 @@ static GstElementDetails mp3parse_details = {
|
|||
"(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*
|
||||
mp3_src_factory (void)
|
||||
{
|
||||
|
|
@ -115,11 +124,221 @@ gst_mp3parse_get_type(void) {
|
|||
0,
|
||||
(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;
|
||||
}
|
||||
|
||||
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
|
||||
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)));
|
||||
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 */
|
||||
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
|
||||
bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
|
||||
{
|
||||
int layer_index,layer,lsf,samplerate_index,padding,mode;
|
||||
long bpf;
|
||||
gint channels, rate;
|
||||
guint bitrate, layer, rate, channels, length;
|
||||
|
||||
/*mpegver = (header >> 19) & 0x3; // don't need this for bpf */
|
||||
layer_index = (header >> 17) & 0x3;
|
||||
layer = 4 - layer_index;
|
||||
lsf = (header & (1 << 20)) ? ((header & (1 << 19)) ? 0 : 1) : 1;
|
||||
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;
|
||||
if (!(length = mp3_type_frame_length_from_header (header, &layer,
|
||||
&channels,
|
||||
&bitrate, &rate))) {
|
||||
return 0;
|
||||
}
|
||||
|
||||
channels = (mode == 3) ? 1 : 2;
|
||||
rate = mp3parse_freqs[samplerate_index];
|
||||
if (channels != parse->channels ||
|
||||
rate != parse->rate ||
|
||||
layer != parse->layer) {
|
||||
GstCaps *caps = GST_CAPS_NEW ("mp3parse_src",
|
||||
"audio/mpeg",
|
||||
"mpegversion", GST_PROPS_INT (1),
|
||||
"layer", GST_PROPS_INT (layer),
|
||||
"channels", GST_PROPS_INT (channels),
|
||||
"rate", GST_PROPS_INT (rate));
|
||||
layer != parse->layer ||
|
||||
bitrate != parse->bit_rate) {
|
||||
GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
|
||||
|
||||
if (gst_pad_try_set_caps(parse->srcpad, caps) <= 0) {
|
||||
gst_element_error (GST_ELEMENT (parse),
|
||||
"mp3parse: failed to negotiate format with next element");
|
||||
|
|
@ -367,12 +556,10 @@ bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
|
|||
parse->channels = channels;
|
||||
parse->layer = layer;
|
||||
parse->rate = rate;
|
||||
parse->bit_rate = bitrate;
|
||||
}
|
||||
|
||||
/*g_print("%08x: layer %d lsf %d bitrate %d samplerate_index %d padding %d - bpf %d\n", */
|
||||
/*header,layer,lsf,bitrate,samplerate_index,padding,bpf); */
|
||||
|
||||
return bpf;
|
||||
return length;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
|
|
@ -470,6 +657,7 @@ static gboolean
|
|||
plugin_init (GModule *module, GstPlugin *plugin)
|
||||
{
|
||||
GstElementFactory *factory;
|
||||
GstTypeFactory *type;
|
||||
|
||||
/* create an elementfactory for the mp3parse element */
|
||||
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));
|
||||
|
||||
/* type finding */
|
||||
type = gst_type_factory_new (&mp3type_definition);
|
||||
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -1235,10 +1235,6 @@ gst_mpeg_demux_plugin_init (GModule *module, GstPlugin *plugin)
|
|||
{
|
||||
GstElementFactory *factory;
|
||||
|
||||
/* this filter needs the bytestream package */
|
||||
if (!gst_library_load ("gstbytestream"))
|
||||
return FALSE;
|
||||
|
||||
/* create an elementfactory for the mpeg_demux element */
|
||||
factory = gst_element_factory_new ("mpegdemux", GST_TYPE_MPEG_DEMUX,
|
||||
&mpeg_demux_details);
|
||||
|
|
|
|||
|
|
@ -24,7 +24,7 @@
|
|||
|
||||
#include <config.h>
|
||||
#include <gst/gst.h>
|
||||
#include <gst/bytestream/bytestream.h>
|
||||
#include <gst/gstbytestream.h>
|
||||
|
||||
|
||||
#ifdef __cplusplus
|
||||
|
|
|
|||
|
|
@ -915,10 +915,6 @@ gst_mpeg_parse_plugin_init (GModule *module, GstPlugin *plugin)
|
|||
{
|
||||
GstElementFactory *factory;
|
||||
|
||||
/* this filter needs the bytestream package */
|
||||
if (!gst_library_load ("gstbytestream"))
|
||||
return FALSE;
|
||||
|
||||
/* create an elementfactory for the mpeg_parse element */
|
||||
factory = gst_element_factory_new ("mpegparse", GST_TYPE_MPEG_PARSE,
|
||||
&mpeg_parse_details);
|
||||
|
|
|
|||
|
|
@ -22,7 +22,7 @@
|
|||
#define __MPEG_PARSE_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/bytestream/bytestream.h>
|
||||
#include <gst/gstbytestream.h>
|
||||
#include "gstmpegpacketize.h"
|
||||
|
||||
#ifdef __cplusplus
|
||||
|
|
|
|||
|
|
@ -25,10 +25,6 @@
|
|||
static gboolean
|
||||
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
|
||||
* 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
|
||||
|
|
|
|||
|
|
@ -328,10 +328,6 @@ gst_rfc2250_enc_plugin_init (GModule *module, GstPlugin *plugin)
|
|||
{
|
||||
GstElementFactory *factory;
|
||||
|
||||
/* this filter needs the bytestream package */
|
||||
if (!gst_library_load("gstbytestream"))
|
||||
return FALSE;
|
||||
|
||||
/* create an elementfactory for the rfc2250_enc element */
|
||||
factory = gst_element_factory_new ("rfc2250enc", GST_TYPE_RFC2250_ENC,
|
||||
&rfc2250_enc_details);
|
||||
|
|
|
|||
|
|
@ -24,7 +24,7 @@
|
|||
|
||||
#include <config.h>
|
||||
#include <gst/gst.h>
|
||||
#include <gst/bytestream/bytestream.h>
|
||||
#include <gst/gstbytestream.h>
|
||||
#include "gstmpegpacketize.h"
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -87,7 +87,7 @@ gst_rmdemux_details =
|
|||
"(C) 2003",
|
||||
};
|
||||
|
||||
static GstCaps* realmedia_type_find (GstBuffer *buf, gpointer private);
|
||||
static GstCaps* realmedia_type_find (GstByteStream *bs, gpointer private);
|
||||
|
||||
static GstTypeDefinition realmediadefinition = {
|
||||
"rmdemux_video/realmedia",
|
||||
|
|
@ -195,21 +195,26 @@ gst_rmdemux_init (GstRMDemux *rmdemux)
|
|||
}
|
||||
|
||||
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_BUFFER_SIZE(buf) < 4){
|
||||
return NULL;
|
||||
if (gst_bytestream_peek (bs, &buf, 4) == 4) {
|
||||
gchar *data = GST_BUFFER_DATA (buf);
|
||||
|
||||
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",
|
||||
"application/vnd.rn-realmedia",
|
||||
NULL);
|
||||
|
||||
if (buf != NULL) {
|
||||
gst_buffer_unref (buf);
|
||||
}
|
||||
return NULL;
|
||||
|
||||
return new;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
|
|
@ -218,9 +223,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
|
|||
GstElementFactory *factory;
|
||||
GstTypeFactory *type;
|
||||
|
||||
if (!gst_library_load ("gstbytestream"))
|
||||
return FALSE;
|
||||
|
||||
factory = gst_element_factory_new ("rmdemux", GST_TYPE_RMDEMUX,
|
||||
&gst_rmdemux_details);
|
||||
g_return_val_if_fail(factory != NULL, FALSE);
|
||||
|
|
|
|||
|
|
@ -22,7 +22,7 @@
|
|||
#define __GST_RMDEMUX_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/bytestream/bytestream.h>
|
||||
#include <gst/gstbytestream.h>
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
|
|
|
|||
Loading…
Reference in a new issue