diff options
Diffstat (limited to 'gst/mpegaudioparse')
| -rw-r--r-- | gst/mpegaudioparse/Makefile.am | 16 | ||||
| -rw-r--r-- | gst/mpegaudioparse/README | 12 | ||||
| -rw-r--r-- | gst/mpegaudioparse/gstmp3types.c | 77 | ||||
| -rw-r--r-- | gst/mpegaudioparse/gstmpegaudioparse.c | 506 | ||||
| -rw-r--r-- | gst/mpegaudioparse/gstmpegaudioparse.h | 71 |
5 files changed, 682 insertions, 0 deletions
diff --git a/gst/mpegaudioparse/Makefile.am b/gst/mpegaudioparse/Makefile.am new file mode 100644 index 00000000..9d670bd3 --- /dev/null +++ b/gst/mpegaudioparse/Makefile.am @@ -0,0 +1,16 @@ +#FIXME clean me up a bit + +filterdir = $(libdir)/gst +filter_LTLIBRARIES = libgstmpegaudioparse.la libgstmp3types.la + +libgstmpegaudioparse_la_SOURCES = gstmpegaudioparse.c gstmp3types.c +# FIXME is this useful? +libgstmpegaudioparse_la_CFLAGS = -O3 $(FOMIT_FRAME_POINTER) -ffast-math -finline-functions $(GST_CFLAGS) + +libgstmp3types_la_SOURCES = gstmp3types.c +libgstmp3types_la_CFLAGS = -O3 $(FOMIT_FRAME_POINTER) -ffast-math -finline-functions $(GST_CFLAGS) + +noinst_HEADERS = gstmpegaudioparse.h +EXTRA_DIST = README + +# FIXME is this needed? diff --git a/gst/mpegaudioparse/README b/gst/mpegaudioparse/README new file mode 100644 index 00000000..8a803d3e --- /dev/null +++ b/gst/mpegaudioparse/README @@ -0,0 +1,12 @@ +MP3 Audio Parser +================ + +This element acts as a parser for mpeg audio data. It's called 'mp3' but +in reality will work for any MPEG-1, MPEG-2, or MPEG-2.5 elemental audio +stream of any of Layers I, II, and III. It will not (currently, ever?) +handle MPEG-2 BC or NBC streams, as those have rather specialized needs +best served be a different filter. + +It will take an mpeg audio stream in any form on its 'src' input, with any +buffer size, and split it into buffers containing a single frame each. +NOTE: ancillary data is not dealt with right now. diff --git a/gst/mpegaudioparse/gstmp3types.c b/gst/mpegaudioparse/gstmp3types.c new file mode 100644 index 00000000..efb9fd5b --- /dev/null +++ b/gst/mpegaudioparse/gstmp3types.c @@ -0,0 +1,77 @@ +/* Gnome-Streamer + * 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> + +static GstCaps* mp3_typefind(GstBuffer *buf, gpointer private); + +static GstTypeDefinition mp3type_definitions[] = { + { "mp3types_audio/mp3", "audio/mp3", ".mp3 .mp2 .mp1 .mpga", mp3_typefind }, + { NULL, NULL, NULL, NULL }, +}; + +static GstCaps* +mp3_typefind(GstBuffer *buf, gpointer private) +{ + gulong head = GULONG_FROM_BE(*((gulong *)GST_BUFFER_DATA(buf))); + GstCaps *caps; + + GST_DEBUG (0,"mp3typefind: typefind\n"); + if ((head & 0xffe00000) != 0xffe00000) + return NULL; + if (!((head >> 17) & 3)) + return NULL; + 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_typefind", "audio/mp3", NULL); +// gst_caps_set(caps,"layer",GST_PROPS_INT(4-((head>>17)&0x3))); + + return caps; +} + +static gboolean +plugin_init (GModule *module, GstPlugin *plugin) +{ + gint i=0; + + while (mp3type_definitions[i].name) { + GstTypeFactory *type; + + type = gst_typefactory_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 new file mode 100644 index 00000000..b1431c73 --- /dev/null +++ b/gst/mpegaudioparse/gstmpegaudioparse.c @@ -0,0 +1,506 @@ +/* Gnome-Streamer + * 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 GST_DEBUG_ENABLED +#include <gstmpegaudioparse.h> + + +/* elementfactory information */ +static GstElementDetails mp3parse_details = { + "MP3 Parser", + "Filter/Parser/Audio", + "Parses and frames MP3 audio streams, provides seek", + VERSION, + "Erik Walthinsen <omega@cse.ogi.edu>", + "(C) 1999", +}; + +static GstPadTemplate* +mp3_src_factory (void) +{ + return + gst_padtemplate_new ( + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + gst_caps_new ( + "mp3parse_src", + "audio/mp3", + gst_props_new ( + "layer", GST_PROPS_INT_RANGE (1, 3), + "bitrate", GST_PROPS_INT_RANGE (8, 320), + "framed", GST_PROPS_BOOLEAN (TRUE), + NULL)), + NULL); +} + +static GstPadTemplate* +mp3_sink_factory (void) +{ + return + gst_padtemplate_new ( + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + gst_caps_new ( + "mp3parse_sink", + "audio/mp3", + NULL), + NULL); +}; + +/* GstMPEGAudioParse signals and args */ +enum { + /* FILL ME */ + LAST_SIGNAL +}; + +enum { + ARG_0, + ARG_SKIP, + ARG_BIT_RATE, + /* FILL ME */ +}; + +static GstPadTemplate *sink_temp, *src_temp; + +static void gst_mp3parse_class_init (GstMPEGAudioParseClass *klass); +static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse); + +static void gst_mp3parse_loop (GstElement *element); +static void gst_mp3parse_chain (GstPad *pad,GstBuffer *buf); +static long bpf_from_header (GstMPEGAudioParse *parse, unsigned long header); +static int head_check (unsigned long head); + +static void gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); +static void gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); + +static GstElementClass *parent_class = NULL; +//static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; + +GType +mp3parse_get_type(void) { + static GType mp3parse_type = 0; + + if (!mp3parse_type) { + static const GTypeInfo mp3parse_info = { + sizeof(GstMPEGAudioParseClass), NULL, + NULL, + (GClassInitFunc)gst_mp3parse_class_init, + NULL, + NULL, + sizeof(GstMPEGAudioParse), + 0, + (GInstanceInitFunc)gst_mp3parse_init, + }; + mp3parse_type = g_type_register_static(GST_TYPE_ELEMENT, "GstMPEGAudioParse", &mp3parse_info, 0); + } + return mp3parse_type; +} + +static void +gst_mp3parse_class_init (GstMPEGAudioParseClass *klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass*)klass; + gstelement_class = (GstElementClass*)klass; + + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_SKIP, + g_param_spec_int("skip","skip","skip", + G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE, + g_param_spec_int("bit_rate","bit_rate","bit_rate", + G_MININT,G_MAXINT,0,G_PARAM_READABLE)); // CHECKME + + parent_class = g_type_class_ref(GST_TYPE_ELEMENT); + + gobject_class->set_property = gst_mp3parse_set_property; + gobject_class->get_property = gst_mp3parse_get_property; +} + +static void +gst_mp3parse_init (GstMPEGAudioParse *mp3parse) +{ + mp3parse->sinkpad = gst_pad_new_from_template(sink_temp, "sink"); + gst_pad_set_caps(mp3parse->sinkpad, gst_pad_get_padtemplate_caps (mp3parse->sinkpad)); + gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->sinkpad); +// gst_pad_set_type_id(mp3parse->sinkpad, mp3type); + +#if 1 // set this to one to use the old chaining code + gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain); +#else // else you get the new loop-based code, which isn't complete yet + gst_element_set_loop_function (GST_ELEMENT(mp3parse),gst_mp3parse_loop); +#endif + + mp3parse->srcpad = gst_pad_new_from_template(src_temp, "src"); + gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad); + //gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); + + mp3parse->partialbuf = NULL; + mp3parse->skip = 0; + mp3parse->in_flush = FALSE; +} + +static guint32 +gst_mp3parse_next_header (guchar *buf,guint32 len,guint32 start) +{ + guint32 offset = start; + int f = 0; + + while (offset < (len - 4)) { + fprintf(stderr,"%02x ",buf[offset]); + if (buf[offset] == 0xff) + f = 1; + else if (f && ((buf[offset] >> 4) == 0x0f)) + return offset - 1; + else + f = 0; + offset++; + } + return -1; +} + +static void +gst_mp3parse_loop (GstElement *element) +{ + GstMPEGAudioParse *parse = GST_MP3PARSE(element); + GstBuffer *inbuf, *outbuf; + guint32 size, offset; + guchar *data; + guint32 start; + guint32 header; + gint bpf; + + while (1) { + // get a new buffer + inbuf = gst_pad_pull (parse->sinkpad); + size = GST_BUFFER_SIZE (inbuf); + data = GST_BUFFER_DATA (inbuf); + offset = 0; +fprintf(stderr, "have buffer of %d bytes\n",size); + + // loop through it and find all the frames + while (offset < (size - 4)) { + start = gst_mp3parse_next_header (data,size,offset); +fprintf(stderr, "skipped %d bytes searching for the next header\n",start-offset); + header = GULONG_FROM_BE(*((guint32 *)(data+start))); +fprintf(stderr, "header is 0x%08x\n",header); + + // figure out how big the frame is supposed to be + bpf = bpf_from_header (parse, header); + + // see if there are enough bytes in this buffer for the whole frame + if ((start + bpf) <= size) { + outbuf = gst_buffer_create_sub (inbuf,start,bpf); +fprintf(stderr, "sending buffer of %d bytes\n",bpf); + gst_pad_push (parse->srcpad, outbuf); + offset = start + bpf; + + // if not, we have to deal with it somehow + } else { +fprintf(stderr,"don't have enough data for this frame\n"); + + break; + } + } + } +} + +static void +gst_mp3parse_chain (GstPad *pad, GstBuffer *buf) +{ + GstMPEGAudioParse *mp3parse; + guchar *data; + glong size,offset = 0; + unsigned long header; + int bpf; + GstBuffer *outbuf; + guint64 last_ts; + + g_return_if_fail(pad != NULL); + g_return_if_fail(GST_IS_PAD(pad)); + g_return_if_fail(buf != NULL); +// g_return_if_fail(GST_IS_BUFFER(buf)); + + mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); + + GST_DEBUG (0,"mp3parse: received buffer of %d bytes\n",GST_BUFFER_SIZE(buf)); + + last_ts = GST_BUFFER_TIMESTAMP(buf); + + if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLUSH)) { + if (mp3parse->partialbuf) { + gst_buffer_unref(mp3parse->partialbuf); + mp3parse->partialbuf = NULL; + } + mp3parse->in_flush = TRUE; + } + + // if we have something left from the previous frame + if (mp3parse->partialbuf) { + + mp3parse->partialbuf = gst_buffer_append(mp3parse->partialbuf, buf); + // and the one we received.. + gst_buffer_unref(buf); + } + else { + mp3parse->partialbuf = buf; + } + + size = GST_BUFFER_SIZE(mp3parse->partialbuf); + data = GST_BUFFER_DATA(mp3parse->partialbuf); + + // while we still have bytes left -4 for the header + while (offset < size-4) { + int skipped = 0; + + GST_DEBUG (0,"mp3parse: offset %ld, size %ld \n",offset, size); + + // search for a possible start byte + for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++; + if (skipped && !mp3parse->in_flush) { + GST_DEBUG (0,"mp3parse: **** now at %ld skipped %d bytes\n",offset,skipped); + } + // construct the header word + header = GULONG_FROM_BE(*((gulong *)(data+offset))); + // if it's a valid header, go ahead and send off the frame + if (head_check(header)) { + // calculate the bpf of the frame + bpf = bpf_from_header(mp3parse, header); + + /******************************************************************************** + * robust seek support + * - This performs additional frame validation if the in_flush flag is set + * (indicating a discontinuous stream). + * - The current frame header is not accepted as valid unless the NEXT frame + * header has the same values for most fields. This significantly increases + * the probability that we aren't processing random data. + * - It is not clear if this is sufficient for robust seeking of Layer III + * streams which utilize the concept of a "bit reservoir" by borrow bitrate + * from previous frames. In this case, seeking may be more complicated because + * the frames are not independently coded. + ********************************************************************************/ + if ( mp3parse->in_flush ) { + unsigned long header2; + + if ((size-offset)<(bpf+4)) { if (mp3parse->in_flush) break; } // wait until we have the the entire current frame as well as the next frame header + + header2 = GULONG_FROM_BE(*((gulong *)(data+offset+bpf))); + GST_DEBUG(0,"mp3parse: header=%08lX, header2=%08lX, bpf=%d\n", header, header2, bpf ); + + #define HDRMASK ~( (0xF<<12)/*bitrate*/ | (1<<9)/*padding*/ | (3<<4)/*mode extension*/ ) // mask the bits which are allowed to differ between frames + + if ( (header2&HDRMASK) != (header&HDRMASK) ) { // require 2 matching headers in a row + GST_DEBUG(0,"mp3parse: next header doesn't match (header=%08lX, header2=%08lX, bpf=%d)\n", header, header2, bpf ); + offset++; // This frame is invalid. Start looking for a valid frame at the next position in the stream + continue; + } + + } + + // if we don't have the whole frame... + if ((size - offset) < bpf) { + GST_DEBUG (0,"mp3parse: partial buffer needed %ld < %d \n",(size-offset), bpf); + break; + } else { + + outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf); + + offset += bpf; + if (mp3parse->skip == 0) { + GST_DEBUG (0,"mp3parse: pushing buffer of %d bytes\n",GST_BUFFER_SIZE(outbuf)); + if (mp3parse->in_flush) { + GST_BUFFER_FLAG_SET(outbuf, GST_BUFFER_FLUSH); + mp3parse->in_flush = FALSE; + } + else { + GST_BUFFER_FLAG_UNSET(outbuf, GST_BUFFER_FLUSH); + } + GST_BUFFER_TIMESTAMP(outbuf) = last_ts; + gst_pad_push(mp3parse->srcpad,outbuf); + } + else { + GST_DEBUG (0,"mp3parse: skipping buffer of %d bytes\n",GST_BUFFER_SIZE(outbuf)); + gst_buffer_unref(outbuf); + mp3parse->skip--; + } + } + } else { + offset++; + if (!mp3parse->in_flush) GST_DEBUG (0,"mp3parse: *** wrong header, skipping byte (FIXME?)\n"); + } + } + // if we have processed this block and there are still + // bytes left not in a partial block, copy them over. + if (size-offset > 0) { + glong remainder = (size - offset); + GST_DEBUG (0,"mp3parse: partial buffer needed %ld for trailing bytes\n",remainder); + + outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,remainder); + gst_buffer_unref(mp3parse->partialbuf); + mp3parse->partialbuf = outbuf; + } + else { + gst_buffer_unref(mp3parse->partialbuf); + mp3parse->partialbuf = NULL; + } +} + +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; + long bpf; + + //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; + + 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; + } + + //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; +} + +static gboolean +head_check (unsigned long head) +{ + GST_DEBUG (0,"checking mp3 header 0x%08lx\n",head); + /* if it's not a valid sync */ + if ((head & 0xffe00000) != 0xffe00000) { + GST_DEBUG (0,"invalid sync\n");return FALSE; } + /* if it's an invalid MPEG version */ + if (((head >> 19) & 3) == 0x1) { + GST_DEBUG (0,"invalid MPEG version\n");return FALSE; } + /* if it's an invalid layer */ + if (!((head >> 17) & 3)) { + GST_DEBUG (0,"invalid layer\n");return FALSE; } + /* if it's an invalid bitrate */ + if (((head >> 12) & 0xf) == 0x0) { + GST_DEBUG (0,"invalid bitrate\n");return FALSE; } + if (((head >> 12) & 0xf) == 0xf) { + GST_DEBUG (0,"invalid bitrate\n");return FALSE; } + /* if it's an invalid samplerate */ + if (((head >> 10) & 0x3) == 0x3) { + GST_DEBUG (0,"invalid samplerate\n");return FALSE; } + if ((head & 0xffff0000) == 0xfffe0000) { + GST_DEBUG (0,"invalid sync\n");return FALSE; } + if (head & 0x00000002) { + GST_DEBUG (0,"invalid emphasis\n");return FALSE; } + + return TRUE; +} + +static void +gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstMPEGAudioParse *src; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail(GST_IS_MP3PARSE(object)); + src = GST_MP3PARSE(object); + + switch (prop_id) { + case ARG_SKIP: + src->skip = g_value_get_int (value); + break; + default: + break; + } +} + +static void +gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +{ + GstMPEGAudioParse *src; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail(GST_IS_MP3PARSE(object)); + src = GST_MP3PARSE(object); + + switch (prop_id) { + case ARG_SKIP: + g_value_set_int (value, src->skip); + break; + case ARG_BIT_RATE: + g_value_set_int (value, src->bit_rate * 1000); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static gboolean +plugin_init (GModule *module, GstPlugin *plugin) +{ + GstElementFactory *factory; + + /* create an elementfactory for the mp3parse element */ + factory = gst_elementfactory_new ("mp3parse", + GST_TYPE_MP3PARSE, + &mp3parse_details); + g_return_val_if_fail (factory != NULL, FALSE); + + sink_temp = mp3_sink_factory (); + gst_elementfactory_add_padtemplate (factory, sink_temp); + + src_temp = mp3_src_factory (); + gst_elementfactory_add_padtemplate (factory, src_temp); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + return TRUE; +} + +GstPluginDesc plugin_desc = { + GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "mp3parse", + plugin_init +}; diff --git a/gst/mpegaudioparse/gstmpegaudioparse.h b/gst/mpegaudioparse/gstmpegaudioparse.h new file mode 100644 index 00000000..fbd1047e --- /dev/null +++ b/gst/mpegaudioparse/gstmpegaudioparse.h @@ -0,0 +1,71 @@ +/* Gnome-Streamer + * 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. + */ + + +#ifndef __MP3PARSE_H__ +#define __MP3PARSE_H__ + + +#include <config.h> +#include <gst/gst.h> + + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + + +#define GST_TYPE_MP3PARSE \ + (gst_mp3parse_get_type()) +#define GST_MP3PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_MP3PARSE,GstMPEGAudioParse)) +#define GST_MP3PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_MP3PARSE,GstMPEGAudioParse)) +#define GST_IS_MP3PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_MP3PARSE)) +#define GST_IS_MP3PARSE_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_MP3PARSE)) + +typedef struct _GstMPEGAudioParse GstMPEGAudioParse; +typedef struct _GstMPEGAudioParseClass GstMPEGAudioParseClass; + +struct _GstMPEGAudioParse { + GstElement element; + + GstPad *sinkpad,*srcpad; + + GstBuffer *partialbuf; // previous buffer (if carryover) + guint skip; /* number of frames to skip */ + guint bit_rate; + gboolean in_flush; +}; + +struct _GstMPEGAudioParseClass { + GstElementClass parent_class; +}; + +GType gst_mp3parse_get_type(void); + + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __MP3PARSE_H__ */ |