diff options
-rw-r--r-- | configure.ac | 3 | ||||
-rw-r--r-- | ext/Makefile.am | 5 | ||||
-rw-r--r-- | ext/audiofile/gstafparse.c | 3 | ||||
-rw-r--r-- | ext/audiofile/gstafparse.h | 2 | ||||
-rw-r--r-- | ext/ivorbis/vorbis.c | 6 | ||||
-rw-r--r-- | ext/ivorbis/vorbisfile.c | 2 | ||||
-rw-r--r-- | ext/jack/gstjack.h | 2 | ||||
-rw-r--r-- | ext/ladspa/gstladspa.c | 3 | ||||
-rw-r--r-- | ext/ladspa/gstladspa.h | 2 | ||||
-rw-r--r-- | ext/mplex/gstmplex.h | 2 | ||||
-rw-r--r-- | gst/cdxaparse/gstcdxaparse.c | 35 | ||||
-rw-r--r-- | gst/cdxaparse/gstcdxaparse.h | 2 | ||||
-rw-r--r-- | gst/festival/gstfestival.c | 36 | ||||
-rw-r--r-- | gst/flx/gstflxdec.c | 38 | ||||
-rw-r--r-- | gst/flx/gstflxdec.h | 2 | ||||
-rw-r--r-- | gst/mixmatrix/mixmatrix.c | 6 | ||||
-rw-r--r-- | gst/modplug/gstmodplug.cc | 56 | ||||
-rw-r--r-- | gst/modplug/gstmodplug.h | 2 | ||||
-rw-r--r-- | gst/mpegaudioparse/Makefile.am | 10 | ||||
-rw-r--r-- | gst/mpegaudioparse/gstmp3types.c | 251 | ||||
-rw-r--r-- | gst/mpegaudioparse/gstmpegaudioparse.c | 288 | ||||
-rw-r--r-- | gst/qtdemux/qtdemux.c | 40 | ||||
-rw-r--r-- | gst/qtdemux/qtdemux.h | 2 |
23 files changed, 373 insertions, 425 deletions
diff --git a/configure.ac b/configure.ac index bf0338ee..5bcb93ac 100644 --- a/configure.ac +++ b/configure.ac @@ -262,7 +262,7 @@ GST_PLUGINS_ALL="\ ac3parse adder audioscale auparse avi \ asfdemux audioconvert cdxaparse chart \ cutter debug deinterlace effectv festival \ - filter flx goom intfloat law level median mixmatrix \ + filter flx goom id3 intfloat law level median mixmatrix \ mpeg1sys mpeg1videoparse mpeg2enc mpeg2sub \ mpegaudio mpegaudioparse mpegstream mpegtypes \ monoscope oneton overlay passthrough playondemand qtdemux \ @@ -1180,6 +1180,7 @@ gst/festival/Makefile gst/filter/Makefile gst/flx/Makefile gst/goom/Makefile +gst/id3/Makefile gst/intfloat/Makefile gst/law/Makefile gst/level/Makefile diff --git a/ext/Makefile.am b/ext/Makefile.am index 2b1ab6ec..66346eac 100644 --- a/ext/Makefile.am +++ b/ext/Makefile.am @@ -278,8 +278,9 @@ SUBDIRS=$(A52DEC_DIR) $(AALIB_DIR) $(ALSA_DIR) \ $(ARTS_DIR) $(ARTSC_DIR) $(AUDIOFILE_DIR) \ $(CDPARANOIA_DIR) $(DIVX_DIR) \ $(DVDREAD_DIR) $(DVDNAV_DIR) $(ESD_DIR) $(MAS_DIR) \ - $(FFMPEG_DIR) $(FLAC_DIR) $(GDK_PIXBUF_DIR) $(GNOMEVFS_DIR) $(GSM_DIR) \ - $(HERMES_DIR) $(JACK_DIR) $(JPEG_DIR) \ + $(FFMPEG_DIR) $(FLAC_DIR) $(GDK_PIXBUF_DIR) \ + $(GNOMEVFS_DIR) $(GSM_DIR) $(HERMES_DIR) \ + $(JACK_DIR) $(JPEG_DIR) \ $(LADSPA_DIR) $(LAME_DIR) $(LCS_DIR) \ $(LIBDV_DIR) $(LIBFAME_DIR) $(LIBPNG_DIR) \ $(MAD_DIR) $(MATROSKA_DIR) $(MIKMOD_DIR) \ diff --git a/ext/audiofile/gstafparse.c b/ext/audiofile/gstafparse.c index 255247e8..183f5147 100644 --- a/ext/audiofile/gstafparse.c +++ b/ext/audiofile/gstafparse.c @@ -337,8 +337,7 @@ gst_afparse_plugin_init (GModule *module, GstPlugin *plugin) /* load audio support library */ if (!gst_library_load ("gstaudio")) return FALSE; - if (!gst_library_load ("gstbytestream")) - return FALSE; + return TRUE; } diff --git a/ext/audiofile/gstafparse.h b/ext/audiofile/gstafparse.h index 9483f248..1cf7b1e1 100644 --- a/ext/audiofile/gstafparse.h +++ b/ext/audiofile/gstafparse.h @@ -27,7 +27,7 @@ #include <config.h> #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #include <audiofile.h> /* what else are we to do */ #include <af_vfs.h> diff --git a/ext/ivorbis/vorbis.c b/ext/ivorbis/vorbis.c index 65ef4b06..35919059 100644 --- a/ext/ivorbis/vorbis.c +++ b/ext/ivorbis/vorbis.c @@ -21,7 +21,7 @@ #include <gst/gst.h> #include <tremor/ivorbiscodec.h> #include <tremor/ivorbisfile.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> extern GType ivorbisfile_get_type(void); @@ -132,10 +132,6 @@ plugin_init (GModule *module, GstPlugin *plugin) gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (file)); - /* this filter needs the bytestream package */ - if (!gst_library_load ("gstbytestream")) - return FALSE; - type = gst_type_factory_new (&vorbisdefinition); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type)); diff --git a/ext/ivorbis/vorbisfile.c b/ext/ivorbis/vorbisfile.c index 86c7baf2..5b270fb9 100644 --- a/ext/ivorbis/vorbisfile.c +++ b/ext/ivorbis/vorbisfile.c @@ -24,7 +24,7 @@ #include <gst/gst.h> #include <tremor/ivorbiscodec.h> #include <tremor/ivorbisfile.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #define GST_TYPE_IVORBISFILE \ (ivorbisfile_get_type()) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 87d7d611..80fd2b63 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -22,7 +22,7 @@ #include <jack/jack.h> #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> //#define JACK_DEBUG(str, a...) g_message (str, ##a) #define JACK_DEBUG(str, a...) diff --git a/ext/ladspa/gstladspa.c b/ext/ladspa/gstladspa.c index 460bdf12..5f16dbf5 100644 --- a/ext/ladspa/gstladspa.c +++ b/ext/ladspa/gstladspa.c @@ -993,9 +993,6 @@ plugin_init (GModule *module, GstPlugin *plugin) LADSPAPluginSearch(ladspa_describe_plugin); - if (! gst_library_load ("gstbytestream")) - return FALSE; - /* initialize dparam support library */ gst_control_init(NULL,NULL); diff --git a/ext/ladspa/gstladspa.h b/ext/ladspa/gstladspa.h index 89a039e9..85348a95 100644 --- a/ext/ladspa/gstladspa.h +++ b/ext/ladspa/gstladspa.h @@ -26,7 +26,7 @@ #include <config.h> #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #include "ladspa.h" diff --git a/ext/mplex/gstmplex.h b/ext/mplex/gstmplex.h index 98b19aca..8b108ed1 100644 --- a/ext/mplex/gstmplex.h +++ b/ext/mplex/gstmplex.h @@ -26,7 +26,7 @@ #include <stdlib.h> #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #include "outputstream.hh" #include "bits.hh" diff --git a/gst/cdxaparse/gstcdxaparse.c b/gst/cdxaparse/gstcdxaparse.c index 5af9bd81..bfe3c8ac 100644 --- a/gst/cdxaparse/gstcdxaparse.c +++ b/gst/cdxaparse/gstcdxaparse.c @@ -50,7 +50,7 @@ static GstElementDetails gst_cdxa_parse_details = { "(C) 2002", }; -static GstCaps* cdxa_type_find (GstBuffer *buf, gpointer private); +static GstCaps* cdxa_type_find (GstByteStream *bs, gpointer private); /* typefactory for 'cdxa' */ static GstTypeDefinition cdxadefinition = { @@ -159,25 +159,28 @@ gst_cdxa_parse_init (GstCDXAParse *cdxa_parse) } static GstCaps* -cdxa_type_find (GstBuffer *buf, - gpointer private) +cdxa_type_find (GstByteStream *bs, + gpointer private) { - gchar *data = GST_BUFFER_DATA (buf); - GstCaps *new; + GstBuffer *buf = NULL; + GstCaps *new = NULL; GST_DEBUG ("cdxa_parse: typefind"); - if (GST_BUFFER_SIZE (buf) < 12) - return NULL; + if (gst_bytestream_peek (bs, &buf, 12) == 12) { + guint32 head1 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[0]), + head2 = GUINT32_FROM_LE (((guint32 *) GST_BUFFER_DATA (buf))[2]); - if (GUINT32_FROM_LE (((guint32 *)data)[0]) != GST_RIFF_TAG_RIFF) - return NULL; - if (GUINT32_FROM_LE (((guint32 *)data)[2]) != GST_RIFF_RIFF_CDXA) - return NULL; + if (head1 == GST_RIFF_TAG_RIFF && head2 == GST_RIFF_RIFF_CDXA) { + new = GST_CAPS_NEW ("cdxa_type_find", + "video/x-cdxa", + NULL); + } + } - new = GST_CAPS_NEW ("cdxa_type_find", - "video/x-cdxa", - NULL); + if (buf != NULL) { + gst_buffer_unref (buf); + } return new; } @@ -342,10 +345,6 @@ plugin_init (GModule *module, GstPlugin *plugin) GstElementFactory *factory; GstTypeFactory *type; - /* this filter needs the riff parser */ - if (!gst_library_load ("gstbytestream")) - return FALSE; - /* create an elementfactory for the cdxa_parse element */ factory = gst_element_factory_new ("cdxaparse", GST_TYPE_CDXA_PARSE, &gst_cdxa_parse_details); diff --git a/gst/cdxaparse/gstcdxaparse.h b/gst/cdxaparse/gstcdxaparse.h index fcf77107..c46df149 100644 --- a/gst/cdxaparse/gstcdxaparse.h +++ b/gst/cdxaparse/gstcdxaparse.h @@ -24,7 +24,7 @@ #include <config.h> #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #ifdef __cplusplus extern "C" { diff --git a/gst/festival/gstfestival.c b/gst/festival/gstfestival.c index 87416184..4113410a 100644 --- a/gst/festival/gstfestival.c +++ b/gst/festival/gstfestival.c @@ -80,7 +80,7 @@ static void gst_festival_class_init (GstFestivalClass *klass); static void gst_festival_init (GstFestival *festival); -static GstCaps* text_type_find (GstBuffer *buf, gpointer private); +static GstCaps* text_type_find (GstByteStream *bs, gpointer private); static void gst_festival_chain (GstPad *pad, GstBuffer *buf); static GstElementStateReturn @@ -198,21 +198,35 @@ gst_festival_init (GstFestival *festival) } static GstCaps* -text_type_find (GstBuffer *buf, gpointer private) +text_type_find (GstByteStream *bs, gpointer private) { - gchar *data = GST_BUFFER_DATA (buf); - gint i; + GstBuffer *buf = NULL; + GstCaps *new = NULL; - /* 20 is arbitrary. 4 is definitely too small. */ - if (GST_BUFFER_SIZE (buf) < 20) - return NULL; +#define TEXT_SIZE 32 - for (i=0; i<GST_BUFFER_SIZE (buf); i++) { - if (!isprint(data[i]) && data[i]!='\n') - return NULL; + /* read arbitrary number and see if it's textual */ + if (gst_bytestream_peek (bs, &buf, TEXT_SIZE) == TEXT_SIZE) { + gchar *data = GST_BUFFER_DATA (buf); + gint i; + + for (i = 0; i < TEXT_SIZE; i++) { + if (!isprint (data[i]) && data[i] != '\n') { + break; + } + } + + /* well, we found something that looks like text... */ + new = gst_caps_new ("text_type_find", + "text/plain", + NULL); + } + + if (buf != NULL) { + gst_buffer_unref (buf); } - return gst_caps_new ("text_type_find", "text/plain", NULL); + return new; } diff --git a/gst/flx/gstflxdec.c b/gst/flx/gstflxdec.c index c20327a7..1f55b7d9 100644 --- a/gst/flx/gstflxdec.c +++ b/gst/flx/gstflxdec.c @@ -28,7 +28,7 @@ #define JIFFIE (GST_SECOND/70) -static GstCaps* flxdec_type_find(GstBuffer *buf, gpointer private); +static GstCaps* flxdec_type_find (GstByteStream *bs, gpointer private); /* flx element information */ static GstElementDetails flxdec_details = { @@ -113,27 +113,33 @@ static void flx_decode_delta_flc (GstFlxDec *, guchar *, guchar *); static GstElementClass *parent_class = NULL; static GstCaps* -flxdec_type_find (GstBuffer *buf, gpointer private) +flxdec_type_find (GstByteStream *bs, gpointer private) { - guchar *data = GST_BUFFER_DATA(buf); - GstCaps *new; + GstBuffer *buf = NULL; + GstCaps *new = NULL; - if (GST_BUFFER_SIZE(buf) < 134){ - return NULL; - } + if (gst_bytestream_peek (bs, &buf, 134) == 134) { + guint8 *data = GST_BUFFER_DATA (buf); - /* check magic */ - if ((data[4] == 0x11 || data[4] == 0x12 - || data[4] == 0x30 || data[4] == 0x44) && data[5] == 0xaf) { + /* check magic */ + if ((data[4] == 0x11 || data[4] == 0x12 || + data[4] == 0x30 || data[4] == 0x44) && + data[5] == 0xaf) { /* check the frame type of the first frame */ if ((data[132] == 0x00 || data[132] == 0xfa) && data[133] == 0xf1) { GST_DEBUG ("GstFlxDec: found supported flx format"); - new = gst_caps_new("flxdec_type_find","video/x-fli", NULL); - return new; + new = gst_caps_new ("flxdec_type_find", + "video/x-fli", + NULL); } + } } - - return NULL; + + if (buf != NULL) { + gst_buffer_unref (buf); + } + + return new; } @@ -684,10 +690,6 @@ plugin_init (GModule *module, GstPlugin *plugin) GstElementFactory *factory; GstTypeFactory *type; - /* this filter needs the bytestream package */ - if (!gst_library_load ("gstbytestream")) - return FALSE; - factory = gst_element_factory_new("flxdec", GST_TYPE_FLXDEC, &flxdec_details); g_return_val_if_fail(factory != NULL, FALSE); gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_PRIMARY); diff --git a/gst/flx/gstflxdec.h b/gst/flx/gstflxdec.h index c7d6f386..0ee4b1c5 100644 --- a/gst/flx/gstflxdec.h +++ b/gst/flx/gstflxdec.h @@ -23,7 +23,7 @@ #include <gst/gst.h> #include "flx_color.h" -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #ifdef __cplusplus diff --git a/gst/mixmatrix/mixmatrix.c b/gst/mixmatrix/mixmatrix.c index 0e9e832a..9e7c7ac8 100644 --- a/gst/mixmatrix/mixmatrix.c +++ b/gst/mixmatrix/mixmatrix.c @@ -22,7 +22,7 @@ #endif #include <config.h> #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #include <gst/audio/audio.h> #include <string.h> @@ -500,10 +500,6 @@ plugin_init (GModule *module, GstPlugin *plugin) { GstElementFactory *factory; - /* this filter needs the bytestream package */ - if (!gst_library_load ("gstbytestream")) - return FALSE; - factory = gst_element_factory_new ("mixmatrix", GST_TYPE_MIXMATRIX, &mixmatrix_details); g_return_val_if_fail (factory != NULL, FALSE); diff --git a/gst/modplug/gstmodplug.cc b/gst/modplug/gstmodplug.cc index 2fa602f7..07a09a1c 100644 --- a/gst/modplug/gstmodplug.cc +++ b/gst/modplug/gstmodplug.cc @@ -130,31 +130,39 @@ static GstElementStateReturn static GstElementClass *parent_class = NULL; static GstCaps* -modplug_type_find (GstBuffer *buf, gpointer priv) -{ - if (GST_BUFFER_SIZE (buf) < 75) - return NULL; - - if (MOD_CheckType (buf) || - Mod_669_CheckType (buf) || - Amf_CheckType (buf) || - Dsm_CheckType (buf) || - Fam_CheckType (buf) || - Gdm_CheckType (buf) || - Imf_CheckType (buf) || - It_CheckType (buf) || - M15_CheckType (buf) || - /*Med_CheckType (buf) || <- FIXME */ - Mtm_CheckType (buf) || - Okt_CheckType (buf) || - S3m_CheckType (buf) || - Xm_CheckType (buf)) { - return gst_caps_new ("modplug_type_find", - "audio/x-mod", - NULL); +modplug_type_find (GstByteStream *bs, gpointer priv) +{ + GstBuffer *buf = NULL; + GstCaps *newc = NULL; + + if (gst_bytestream_peek (bs, &buf, 75) == 75) { + if (MOD_CheckType (buf) || + Mod_669_CheckType (buf) || + Amf_CheckType (buf) || + Dsm_CheckType (buf) || + Fam_CheckType (buf) || + Gdm_CheckType (buf) || + Imf_CheckType (buf) || + It_CheckType (buf) || + M15_CheckType (buf) || +#if 0 + Med_CheckType (buf) || /* FIXME */ +#endif + Mtm_CheckType (buf) || + Okt_CheckType (buf) || + S3m_CheckType (buf) || + Xm_CheckType (buf)) { + newc = GST_CAPS_NEW ("modplug_type_find", + "audio/x-mod", + NULL); + } } - - return NULL; + + if (buf != NULL) { + gst_buffer_unref (buf); + } + + return newc; } static GstTypeDefinition modplug_definitions[] = { diff --git a/gst/modplug/gstmodplug.h b/gst/modplug/gstmodplug.h index 97d6d00e..dcdbf062 100644 --- a/gst/modplug/gstmodplug.h +++ b/gst/modplug/gstmodplug.h @@ -28,7 +28,7 @@ extern "C" { #endif /* __cplusplus */ #include <gst/gst.h> -#include <gst/bytestream/bytestream.h> +#include <gst/gstbytestream.h> #include "modplug_types.h" diff --git a/gst/mpegaudioparse/Makefile.am b/gst/mpegaudioparse/Makefile.am index ee22c43e..cb06727f 100644 --- a/gst/mpegaudioparse/Makefile.am +++ b/gst/mpegaudioparse/Makefile.am @@ -1,17 +1,9 @@ -#FIXME clean me up a bit - -plugin_LTLIBRARIES = libgstmpegaudioparse.la libgstmp3types.la +plugin_LTLIBRARIES = libgstmpegaudioparse.la libgstmpegaudioparse_la_SOURCES = gstmpegaudioparse.c libgstmpegaudioparse_la_CFLAGS = $(GST_CFLAGS) libgstmpegaudioparse_la_LIBADD = libgstmpegaudioparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -libgstmp3types_la_SOURCES = gstmp3types.c -libgstmp3types_la_CFLAGS = $(GST_CFLAGS) -libgstmp3types_la_LIBADD = -libgstmp3types_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) - noinst_HEADERS = gstmpegaudioparse.h EXTRA_DIST = README - diff --git a/gst/mpegaudioparse/gstmp3types.c b/gst/mpegaudioparse/gstmp3types.c deleted file mode 100644 index cf5e860f..00000000 --- a/gst/mpegaudioparse/gstmp3types.c +++ /dev/null @@ -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 -}; diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c index ba207c0e..920ab22c 100644 --- a/gst/mpegaudioparse/gstmpegaudioparse.c +++ b/gst/mpegaudioparse/gstmpegaudioparse.c @@ -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; - - /*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; + guint bitrate, layer, rate, channels, length; - 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; } diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 3358db33..f5a95c03 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -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); - - g_return_val_if_fail (data != NULL, NULL); - - if(GST_BUFFER_SIZE(buf) < 8){ - return NULL; + GstBuffer *buf = NULL; + GstCaps *new = 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); diff --git a/gst/qtdemux/qtdemux.h b/gst/qtdemux/qtdemux.h index 108faab1..46684be8 100644 --- a/gst/qtdemux/qtdemux.h +++ b/gst/qtdemux/qtdemux.h @@ -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" { |