mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +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:
parent
b569848e28
commit
d9e4457faa
23 changed files with 371 additions and 423 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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@ -337,8 +337,7 @@ gst_afparse_plugin_init (GModule *module, GstPlugin *plugin)
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/* load audio support library */
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if (!gst_library_load ("gstaudio"))
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return FALSE;
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if (!gst_library_load ("gstbytestream"))
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return FALSE;
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return TRUE;
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}
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@ -27,7 +27,7 @@
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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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#include <audiofile.h> /* what else are we to do */
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#include <af_vfs.h>
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@ -21,7 +21,7 @@
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#include <gst/gst.h>
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#include <tremor/ivorbiscodec.h>
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#include <tremor/ivorbisfile.h>
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#include <gst/bytestream/bytestream.h>
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#include <gst/gstbytestream.h>
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extern GType ivorbisfile_get_type(void);
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@ -132,10 +132,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
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gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (file));
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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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type = gst_type_factory_new (&vorbisdefinition);
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gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
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@ -24,7 +24,7 @@
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#include <gst/gst.h>
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#include <tremor/ivorbiscodec.h>
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#include <tremor/ivorbisfile.h>
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#include <gst/bytestream/bytestream.h>
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#include <gst/gstbytestream.h>
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#define GST_TYPE_IVORBISFILE \
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(ivorbisfile_get_type())
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@ -22,7 +22,7 @@
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#include <jack/jack.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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//#define JACK_DEBUG(str, a...) g_message (str, ##a)
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#define JACK_DEBUG(str, a...)
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@ -993,9 +993,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
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LADSPAPluginSearch(ladspa_describe_plugin);
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if (! gst_library_load ("gstbytestream"))
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return FALSE;
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/* initialize dparam support library */
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gst_control_init(NULL,NULL);
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@ -26,7 +26,7 @@
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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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#include "ladspa.h"
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@ -26,7 +26,7 @@
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#include <stdlib.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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#include "outputstream.hh"
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#include "bits.hh"
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@ -50,7 +50,7 @@ static GstElementDetails gst_cdxa_parse_details = {
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"(C) 2002",
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};
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static GstCaps* cdxa_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* cdxa_type_find (GstByteStream *bs, gpointer private);
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/* typefactory for 'cdxa' */
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static GstTypeDefinition cdxadefinition = {
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@ -159,25 +159,28 @@ gst_cdxa_parse_init (GstCDXAParse *cdxa_parse)
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}
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static GstCaps*
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cdxa_type_find (GstBuffer *buf,
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gpointer private)
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cdxa_type_find (GstByteStream *bs,
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gpointer private)
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{
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gchar *data = GST_BUFFER_DATA (buf);
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GstCaps *new;
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GstBuffer *buf = NULL;
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GstCaps *new = NULL;
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GST_DEBUG ("cdxa_parse: typefind");
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if (GST_BUFFER_SIZE (buf) < 12)
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return NULL;
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if (gst_bytestream_peek (bs, &buf, 12) == 12) {
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guint32 head1 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[0]),
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head2 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[2]);
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if (GUINT32_FROM_LE (((guint32 *)data)[0]) != GST_RIFF_TAG_RIFF)
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return NULL;
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if (GUINT32_FROM_LE (((guint32 *)data)[2]) != GST_RIFF_RIFF_CDXA)
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return NULL;
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if (head1 == GST_RIFF_TAG_RIFF && head2 == GST_RIFF_RIFF_CDXA) {
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new = GST_CAPS_NEW ("cdxa_type_find",
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"video/x-cdxa",
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NULL);
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}
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}
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new = GST_CAPS_NEW ("cdxa_type_find",
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"video/x-cdxa",
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NULL);
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if (buf != NULL) {
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gst_buffer_unref (buf);
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}
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return new;
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}
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@ -342,10 +345,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
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GstElementFactory *factory;
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GstTypeFactory *type;
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/* this filter needs the riff parser */
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if (!gst_library_load ("gstbytestream"))
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return FALSE;
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/* create an elementfactory for the cdxa_parse element */
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factory = gst_element_factory_new ("cdxaparse", GST_TYPE_CDXA_PARSE,
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&gst_cdxa_parse_details);
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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/bytestream/bytestream.h>
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#include <gst/gstbytestream.h>
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#ifdef __cplusplus
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extern "C" {
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@ -80,7 +80,7 @@
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static void gst_festival_class_init (GstFestivalClass *klass);
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static void gst_festival_init (GstFestival *festival);
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static GstCaps* text_type_find (GstBuffer *buf, gpointer private);
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static GstCaps* text_type_find (GstByteStream *bs, gpointer private);
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static void gst_festival_chain (GstPad *pad, GstBuffer *buf);
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static GstElementStateReturn
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@ -198,21 +198,35 @@ gst_festival_init (GstFestival *festival)
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}
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static GstCaps*
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text_type_find (GstBuffer *buf, gpointer private)
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text_type_find (GstByteStream *bs, gpointer private)
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{
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gchar *data = GST_BUFFER_DATA (buf);
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gint i;
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GstBuffer *buf = NULL;
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GstCaps *new = NULL;
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/* 20 is arbitrary. 4 is definitely too small. */
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if (GST_BUFFER_SIZE (buf) < 20)
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return NULL;
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#define TEXT_SIZE 32
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for (i=0; i<GST_BUFFER_SIZE (buf); i++) {
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if (!isprint(data[i]) && data[i]!='\n')
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return NULL;
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/* read arbitrary number and see if it's textual */
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if (gst_bytestream_peek (bs, &buf, TEXT_SIZE) == TEXT_SIZE) {
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gchar *data = GST_BUFFER_DATA (buf);
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gint i;
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for (i = 0; i < TEXT_SIZE; i++) {
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if (!isprint (data[i]) && data[i] != '\n') {
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break;
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}
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}
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/* well, we found something that looks like text... */
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new = gst_caps_new ("text_type_find",
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"text/plain",
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NULL);
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}
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return gst_caps_new ("text_type_find", "text/plain", NULL);
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if (buf != NULL) {
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gst_buffer_unref (buf);
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}
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return new;
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}
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@ -28,7 +28,7 @@
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#define JIFFIE (GST_SECOND/70)
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static GstCaps* flxdec_type_find(GstBuffer *buf, gpointer private);
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static GstCaps* flxdec_type_find (GstByteStream *bs, gpointer private);
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/* flx element information */
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static GstElementDetails flxdec_details = {
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static GstElementClass *parent_class = NULL;
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static GstCaps*
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flxdec_type_find (GstBuffer *buf, gpointer private)
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flxdec_type_find (GstByteStream *bs, gpointer private)
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{
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guchar *data = GST_BUFFER_DATA(buf);
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GstCaps *new;
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GstBuffer *buf = NULL;
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GstCaps *new = NULL;
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if (GST_BUFFER_SIZE(buf) < 134){
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return NULL;
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}
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if (gst_bytestream_peek (bs, &buf, 134) == 134) {
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guint8 *data = GST_BUFFER_DATA (buf);
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/* check magic */
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if ((data[4] == 0x11 || data[4] == 0x12
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|| data[4] == 0x30 || data[4] == 0x44) && data[5] == 0xaf) {
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/* check magic */
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if ((data[4] == 0x11 || data[4] == 0x12 ||
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data[4] == 0x30 || data[4] == 0x44) &&
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data[5] == 0xaf) {
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/* check the frame type of the first frame */
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if ((data[132] == 0x00 || data[132] == 0xfa) && data[133] == 0xf1) {
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GST_DEBUG ("GstFlxDec: found supported flx format");
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new = gst_caps_new("flxdec_type_find","video/x-fli", NULL);
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return new;
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new = gst_caps_new ("flxdec_type_find",
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"video/x-fli",
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NULL);
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}
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}
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}
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return NULL;
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if (buf != NULL) {
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gst_buffer_unref (buf);
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}
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return new;
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}
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@ -684,10 +690,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
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GstElementFactory *factory;
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GstTypeFactory *type;
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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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factory = gst_element_factory_new("flxdec", GST_TYPE_FLXDEC, &flxdec_details);
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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_PRIMARY);
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@ -23,7 +23,7 @@
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#include <gst/gst.h>
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#include "flx_color.h"
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#include <gst/bytestream/bytestream.h>
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#include <gst/gstbytestream.h>
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#ifdef __cplusplus
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@ -22,7 +22,7 @@
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#endif
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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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#include <gst/audio/audio.h>
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#include <string.h>
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@ -500,10 +500,6 @@ 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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factory = gst_element_factory_new ("mixmatrix", GST_TYPE_MIXMATRIX,
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&mixmatrix_details);
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g_return_val_if_fail (factory != NULL, FALSE);
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@ -130,31 +130,39 @@ static GstElementStateReturn
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static GstElementClass *parent_class = NULL;
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static GstCaps*
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modplug_type_find (GstBuffer *buf, gpointer priv)
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{
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if (GST_BUFFER_SIZE (buf) < 75)
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return NULL;
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modplug_type_find (GstByteStream *bs, gpointer priv)
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{
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GstBuffer *buf = NULL;
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GstCaps *newc = NULL;
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if (MOD_CheckType (buf) ||
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Mod_669_CheckType (buf) ||
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Amf_CheckType (buf) ||
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Dsm_CheckType (buf) ||
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Fam_CheckType (buf) ||
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Gdm_CheckType (buf) ||
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Imf_CheckType (buf) ||
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It_CheckType (buf) ||
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M15_CheckType (buf) ||
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/*Med_CheckType (buf) || <- FIXME */
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Mtm_CheckType (buf) ||
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Okt_CheckType (buf) ||
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S3m_CheckType (buf) ||
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Xm_CheckType (buf)) {
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return gst_caps_new ("modplug_type_find",
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"audio/x-mod",
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NULL);
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if (gst_bytestream_peek (bs, &buf, 75) == 75) {
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if (MOD_CheckType (buf) ||
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Mod_669_CheckType (buf) ||
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Amf_CheckType (buf) ||
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Dsm_CheckType (buf) ||
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Fam_CheckType (buf) ||
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Gdm_CheckType (buf) ||
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Imf_CheckType (buf) ||
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It_CheckType (buf) ||
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M15_CheckType (buf) ||
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#if 0
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Med_CheckType (buf) || /* FIXME */
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#endif
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Mtm_CheckType (buf) ||
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Okt_CheckType (buf) ||
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S3m_CheckType (buf) ||
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Xm_CheckType (buf)) {
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newc = GST_CAPS_NEW ("modplug_type_find",
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"audio/x-mod",
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NULL);
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}
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}
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return NULL;
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if (buf != NULL) {
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gst_buffer_unref (buf);
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}
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return newc;
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}
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static GstTypeDefinition modplug_definitions[] = {
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@ -28,7 +28,7 @@ extern "C" {
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#endif /* __cplusplus */
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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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#include "modplug_types.h"
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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
|
||||
libgstmp3types_la_CFLAGS = $(GST_CFLAGS)
|
||||
libgstmp3types_la_LIBADD =
|
||||
libgstmp3types_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
|
||||
|
||||
noinst_HEADERS = gstmpegaudioparse.h
|
||||
EXTRA_DIST = README
|
||||
|
||||
|
|
|
@ -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
|
||||
};
|
|
@ -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;
|
||||
}
|
||||
|
||||
|
|
|
@ -104,7 +104,7 @@ gst_qtdemux_details =
|
|||
"(C) 2003",
|
||||
};
|
||||
|
||||
static GstCaps* quicktime_type_find (GstBuffer *buf, gpointer private);
|
||||
static GstCaps* quicktime_type_find (GstByteStream *bs, gpointer private);
|
||||
|
||||
static GstTypeDefinition quicktimedefinition = {
|
||||
"qtdemux_video/quicktime",
|
||||
|
@ -187,24 +187,29 @@ gst_qtdemux_init (GstQTDemux *qtdemux)
|
|||
}
|
||||
|
||||
static GstCaps*
|
||||
quicktime_type_find (GstBuffer *buf, gpointer private)
|
||||
quicktime_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) < 8){
|
||||
return NULL;
|
||||
if (gst_bytestream_peek (bs, &buf, 8) == 8) {
|
||||
gchar *data = GST_BUFFER_DATA (buf);
|
||||
|
||||
if (!strncmp (&data[4], "wide", 4) ||
|
||||
!strncmp (&data[4], "moov", 4) ||
|
||||
!strncmp (&data[4], "mdat", 4) ||
|
||||
!strncmp (&data[4], "free", 4)) {
|
||||
new = GST_CAPS_NEW ("quicktime_type_find",
|
||||
"video/quicktime",
|
||||
NULL);
|
||||
}
|
||||
}
|
||||
if (strncmp (&data[4], "wide", 4)==0 ||
|
||||
strncmp (&data[4], "moov", 4)==0 ||
|
||||
strncmp (&data[4], "mdat", 4)==0 ||
|
||||
strncmp (&data[4], "free", 4)==0) {
|
||||
return gst_caps_new ("quicktime_type_find",
|
||||
"video/quicktime",
|
||||
NULL);
|
||||
|
||||
if (buf != NULL) {
|
||||
gst_buffer_unref (buf);
|
||||
}
|
||||
return NULL;
|
||||
|
||||
return new;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
|
@ -222,9 +227,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
|
|||
};
|
||||
gint i;
|
||||
|
||||
if (!gst_library_load ("gstbytestream"))
|
||||
return FALSE;
|
||||
|
||||
factory = gst_element_factory_new ("qtdemux", GST_TYPE_QTDEMUX,
|
||||
&gst_qtdemux_details);
|
||||
g_return_val_if_fail(factory != NULL, FALSE);
|
||||
|
|
|
@ -22,7 +22,7 @@
|
|||
#define __GST_QTDEMUX_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/bytestream/bytestream.h>
|
||||
#include <gst/gstbytestream.h>
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
|
|
Loading…
Reference in a new issue