/* GStreamer * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> * Copyright (C) <2006> Tim-Philipp Müller <tim centricular net> * * 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. */ /** * SECTION:element-auparse * * Parses .au files mostly originating from sun os based computers. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <stdlib.h> #include <string.h> #include "gstauparse.h" #include <gst/audio/audio.h> GST_DEBUG_CATEGORY_STATIC (auparse_debug); #define GST_CAT_DEFAULT (auparse_debug) static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-au") ); #define GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS \ "audio/x-raw, " \ "format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \ "S32LE, S32BE, F32LE, F32BE, " \ "F64LE, F64BE }, " \ "rate = (int) [ 8000, 192000 ], " \ "channels = (int) 1, " \ "layout = (string) interleaved;" \ "audio/x-raw, " \ "format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \ "S32LE, S32BE, F32LE, F32BE, " \ "F64LE, F64BE }, " \ "rate = (int) [ 8000, 192000 ], " \ "channels = (int) 2, " \ "channel-mask = (bitmask) 0x3," \ "layout = (string) interleaved" #define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \ "audio/x-alaw, " \ "rate = (int) [ 8000, 192000 ], " \ "channels = (int) [ 1, 2 ]" #define GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS \ "audio/x-mulaw, " \ "rate = (int) [ 8000, 192000 ], " \ "channels = (int) [ 1, 2 ]" /* Nothing to decode those ADPCM streams for now */ #define GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS \ "audio/x-adpcm, " \ "layout = (string) { g721, g722, g723_3, g723_5 }" static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS "; " GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS ";" GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS ";" GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS)); static void gst_au_parse_dispose (GObject * object); static GstFlowReturn gst_au_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf); static GstStateChangeReturn gst_au_parse_change_state (GstElement * element, GstStateChange transition); static void gst_au_parse_reset (GstAuParse * auparse); static gboolean gst_au_parse_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_au_parse_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_au_parse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_au_parse_src_convert (GstAuParse * auparse, GstFormat src_format, gint64 srcval, GstFormat dest_format, gint64 * destval); #define gst_au_parse_parent_class parent_class G_DEFINE_TYPE (GstAuParse, gst_au_parse, GST_TYPE_ELEMENT); static void gst_au_parse_class_init (GstAuParseClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GST_DEBUG_CATEGORY_INIT (auparse_debug, "auparse", 0, ".au parser"); gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->dispose = gst_au_parse_dispose; gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_au_parse_change_state); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&sink_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&src_template)); gst_element_class_set_static_metadata (gstelement_class, "AU audio demuxer", "Codec/Demuxer/Audio", "Parse an .au file into raw audio", "Erik Walthinsen <omega@cse.ogi.edu>"); } static void gst_au_parse_init (GstAuParse * auparse) { auparse->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink"); gst_pad_set_chain_function (auparse->sinkpad, GST_DEBUG_FUNCPTR (gst_au_parse_chain)); gst_pad_set_event_function (auparse->sinkpad, GST_DEBUG_FUNCPTR (gst_au_parse_sink_event)); gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad); auparse->srcpad = gst_pad_new_from_static_template (&src_template, "src"); gst_pad_set_query_function (auparse->srcpad, GST_DEBUG_FUNCPTR (gst_au_parse_src_query)); gst_pad_set_event_function (auparse->srcpad, GST_DEBUG_FUNCPTR (gst_au_parse_src_event)); gst_pad_use_fixed_caps (auparse->srcpad); gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad); auparse->adapter = gst_adapter_new (); gst_au_parse_reset (auparse); } static void gst_au_parse_dispose (GObject * object) { GstAuParse *au = GST_AU_PARSE (object); if (au->adapter != NULL) { g_object_unref (au->adapter); au->adapter = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_au_parse_reset (GstAuParse * auparse) { auparse->offset = 0; auparse->buffer_offset = 0; auparse->encoding = 0; auparse->samplerate = 0; auparse->channels = 0; gst_adapter_clear (auparse->adapter); /* gst_segment_init (&auparse->segment, GST_FORMAT_TIME); */ } static void gst_au_parse_negotiate_srcpad (GstAuParse * auparse, GstCaps * new_caps) { if (auparse->src_caps && gst_caps_is_equal (new_caps, auparse->src_caps)) { GST_LOG_OBJECT (auparse, "same caps, nothing to do"); return; } gst_caps_replace (&auparse->src_caps, new_caps); GST_DEBUG_OBJECT (auparse, "Changing src pad caps to %" GST_PTR_FORMAT, auparse->src_caps); gst_pad_set_caps (auparse->srcpad, auparse->src_caps); return; } static GstFlowReturn gst_au_parse_parse_header (GstAuParse * auparse) { GstCaps *tempcaps; guint32 size; guint8 *head; gchar layout[7] = { 0, }; GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN; gint law = 0; guint endianness; head = (guint8 *) gst_adapter_map (auparse->adapter, 24); g_assert (head != NULL); GST_DEBUG_OBJECT (auparse, "[%c%c%c%c]", head[0], head[1], head[2], head[3]); switch (GST_READ_UINT32_BE (head)) { /* normal format is big endian (au is a Sparc format) */ case 0x2e736e64:{ /* ".snd" */ endianness = G_BIG_ENDIAN; break; } /* and of course, someone had to invent a little endian * version. Used by DEC systems. */ case 0x646e732e: /* dns. */ case 0x0064732e:{ /* other source say it is "dns." */ endianness = G_LITTLE_ENDIAN; break; } default:{ goto unknown_header; } } auparse->offset = GST_READ_UINT32_BE (head + 4); /* Do not trust size, could be set to -1 : unknown * otherwise: filesize = size + auparse->offset */ size = GST_READ_UINT32_BE (head + 8); auparse->encoding = GST_READ_UINT32_BE (head + 12); auparse->samplerate = GST_READ_UINT32_BE (head + 16); auparse->channels = GST_READ_UINT32_BE (head + 20); if (auparse->samplerate < 8000 || auparse->samplerate > 192000) goto unsupported_sample_rate; if (auparse->channels < 1 || auparse->channels > 2) goto unsupported_number_of_channels; GST_DEBUG_OBJECT (auparse, "offset %" G_GINT64_FORMAT ", size %u, " "encoding %u, frequency %u, channels %u", auparse->offset, size, auparse->encoding, auparse->samplerate, auparse->channels); /* Docs: * http://www.opengroup.org/public/pubs/external/auformat.html * http://astronomy.swin.edu.au/~pbourke/dataformats/au/ * Solaris headers : /usr/include/audio/au.h * libsndfile : src/au.c * * Samples : * http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html */ switch (auparse->encoding) { case 1: /* 8-bit ISDN mu-law G.711 */ law = 1; break; case 27: /* 8-bit ISDN A-law G.711 */ law = 2; break; case 2: /* 8-bit linear PCM, FIXME signed? */ format = GST_AUDIO_FORMAT_S8; auparse->sample_size = auparse->channels; break; case 3: /* 16-bit linear PCM */ if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_S16LE; else format = GST_AUDIO_FORMAT_S16BE; auparse->sample_size = auparse->channels * 2; break; case 4: /* 24-bit linear PCM */ if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_S24LE; else format = GST_AUDIO_FORMAT_S24BE; auparse->sample_size = auparse->channels * 3; break; case 5: /* 32-bit linear PCM */ if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_S32LE; else format = GST_AUDIO_FORMAT_S32BE; auparse->sample_size = auparse->channels * 4; break; case 6: /* 32-bit IEEE floating point */ if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_F32LE; else format = GST_AUDIO_FORMAT_F32BE; auparse->sample_size = auparse->channels * 4; break; case 7: /* 64-bit IEEE floating point */ if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_F64LE; else format = GST_AUDIO_FORMAT_F64BE; auparse->sample_size = auparse->channels * 8; break; case 23: /* 4-bit CCITT G.721 ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */ strcpy (layout, "g721"); break; case 24: /* 8-bit CCITT G.722 ADPCM -> rtp */ strcpy (layout, "g722"); break; case 25: /* 3-bit CCITT G.723.3 ADPCM 24kbps -> rtp/xine/modplug/libsndfile */ strcpy (layout, "g723_3"); break; case 26: /* 5-bit CCITT G.723.5 ADPCM 40kbps -> rtp/xine/modplug/libsndfile */ strcpy (layout, "g723_5"); break; case 8: /* Fragmented sample data */ case 9: /* AU_ENCODING_NESTED */ case 10: /* DSP program */ case 11: /* DSP 8-bit fixed point */ case 12: /* DSP 16-bit fixed point */ case 13: /* DSP 24-bit fixed point */ case 14: /* DSP 32-bit fixed point */ case 16: /* AU_ENCODING_DISPLAY : non-audio display data */ case 17: /* AU_ENCODING_MULAW_SQUELCH */ case 18: /* 16-bit linear with emphasis */ case 19: /* 16-bit linear compressed (NeXT) */ case 20: /* 16-bit linear with emphasis and compression */ case 21: /* Music kit DSP commands */ case 22: /* Music kit DSP commands samples */ default: goto unknown_format; } if (law) { tempcaps = gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw", "rate", G_TYPE_INT, auparse->samplerate, "channels", G_TYPE_INT, auparse->channels, NULL); auparse->sample_size = auparse->channels; } else if (format != GST_AUDIO_FORMAT_UNKNOWN) { GstCaps *templ_caps = gst_pad_get_pad_template_caps (auparse->srcpad); GstCaps *intersection; tempcaps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, gst_audio_format_to_string (format), "rate", G_TYPE_INT, auparse->samplerate, "channels", G_TYPE_INT, auparse->channels, NULL); intersection = gst_caps_intersect (tempcaps, templ_caps); gst_caps_unref (tempcaps); gst_caps_unref (templ_caps); tempcaps = intersection; } else if (layout[0]) { tempcaps = gst_caps_new_simple ("audio/x-adpcm", "layout", G_TYPE_STRING, layout, NULL); auparse->sample_size = 0; } else goto unknown_format; GST_DEBUG_OBJECT (auparse, "sample_size=%d", auparse->sample_size); gst_au_parse_negotiate_srcpad (auparse, tempcaps); GST_DEBUG_OBJECT (auparse, "offset=%" G_GINT64_FORMAT, auparse->offset); gst_adapter_unmap (auparse->adapter); gst_adapter_flush (auparse->adapter, auparse->offset); gst_caps_unref (tempcaps); return GST_FLOW_OK; /* ERRORS */ unknown_header: { gst_adapter_unmap (auparse->adapter); GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL)); return GST_FLOW_ERROR; } unsupported_sample_rate: { gst_adapter_unmap (auparse->adapter); GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), ("Unsupported samplerate: %u", auparse->samplerate)); return GST_FLOW_ERROR; } unsupported_number_of_channels: { gst_adapter_unmap (auparse->adapter); GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), ("Unsupported number of channels: %u", auparse->channels)); return GST_FLOW_ERROR; } unknown_format: { gst_adapter_unmap (auparse->adapter); GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), ("Unsupported encoding: %u", auparse->encoding)); return GST_FLOW_ERROR; } } #define AU_HEADER_SIZE 24 static GstFlowReturn gst_au_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; GstAuParse *auparse; gint avail, sendnow = 0; gint64 timestamp; gint64 duration; gint64 offset; GstSegment segment; auparse = GST_AU_PARSE (parent); GST_LOG_OBJECT (auparse, "got buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (buf)); gst_adapter_push (auparse->adapter, buf); buf = NULL; /* if we haven't seen any data yet... */ if (!gst_pad_has_current_caps (auparse->srcpad)) { if (gst_adapter_available (auparse->adapter) < AU_HEADER_SIZE) { GST_DEBUG_OBJECT (auparse, "need more data to parse header"); ret = GST_FLOW_OK; goto out; } ret = gst_au_parse_parse_header (auparse); if (ret != GST_FLOW_OK) goto out; gst_segment_init (&segment, GST_FORMAT_TIME); gst_pad_push_event (auparse->srcpad, gst_event_new_segment (&segment)); } avail = gst_adapter_available (auparse->adapter); if (auparse->sample_size > 0) { /* Ensure we push a buffer that's a multiple of the frame size downstream */ sendnow = avail - (avail % auparse->sample_size); } else { /* It's something non-trivial (such as ADPCM), we don't understand it, so * just push downstream and assume it will know what to do with it */ sendnow = avail; } if (sendnow > 0) { GstBuffer *outbuf; gint64 pos; outbuf = gst_adapter_take_buffer (auparse->adapter, sendnow); outbuf = gst_buffer_make_writable (outbuf); pos = auparse->buffer_offset - auparse->offset; pos = MAX (pos, 0); if (auparse->sample_size > 0 && auparse->samplerate > 0) { gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos, GST_FORMAT_DEFAULT, &offset); gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos, GST_FORMAT_TIME, ×tamp); gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, sendnow, GST_FORMAT_TIME, &duration); GST_BUFFER_OFFSET (outbuf) = offset; GST_BUFFER_TIMESTAMP (outbuf) = timestamp; GST_BUFFER_DURATION (outbuf) = duration; } auparse->buffer_offset += sendnow; ret = gst_pad_push (auparse->srcpad, outbuf); } out: return ret; } static gboolean gst_au_parse_src_convert (GstAuParse * auparse, GstFormat src_format, gint64 srcval, GstFormat dest_format, gint64 * destval) { gboolean ret = TRUE; guint samplesize, rate; if (dest_format == src_format) { *destval = srcval; return TRUE; } GST_OBJECT_LOCK (auparse); samplesize = auparse->sample_size; rate = auparse->samplerate; GST_OBJECT_UNLOCK (auparse); if (samplesize == 0 || rate == 0) { GST_LOG_OBJECT (auparse, "cannot convert, sample_size or rate unknown"); return FALSE; } switch (src_format) { case GST_FORMAT_BYTES: srcval /= samplesize; /* fallthrough */ case GST_FORMAT_DEFAULT:{ switch (dest_format) { case GST_FORMAT_DEFAULT: *destval = srcval; break; case GST_FORMAT_BYTES: *destval = srcval * samplesize; break; case GST_FORMAT_TIME: *destval = gst_util_uint64_scale_int (srcval, GST_SECOND, rate); break; default: ret = FALSE; break; } break; } case GST_FORMAT_TIME:{ switch (dest_format) { case GST_FORMAT_BYTES: *destval = samplesize * gst_util_uint64_scale_int (srcval, rate, GST_SECOND); break; case GST_FORMAT_DEFAULT: *destval = gst_util_uint64_scale_int (srcval, rate, GST_SECOND); break; default: ret = FALSE; break; } break; } default:{ ret = FALSE; break; } } if (!ret) { GST_DEBUG_OBJECT (auparse, "could not convert from %s to %s format", gst_format_get_name (src_format), gst_format_get_name (dest_format)); } return ret; } static gboolean gst_au_parse_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstAuParse *auparse; gboolean ret = FALSE; auparse = GST_AU_PARSE (parent); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_DURATION:{ GstFormat format; gint64 len, val; gst_query_parse_duration (query, &format, NULL); if (!gst_pad_peer_query_duration (auparse->sinkpad, GST_FORMAT_BYTES, &len)) { GST_DEBUG_OBJECT (auparse, "failed to query upstream length"); break; } GST_OBJECT_LOCK (auparse); len -= auparse->offset; GST_OBJECT_UNLOCK (auparse); ret = gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, len, format, &val); if (ret) { gst_query_set_duration (query, format, val); } break; } case GST_QUERY_POSITION:{ GstFormat format; gint64 pos, val; gst_query_parse_position (query, &format, NULL); if (!gst_pad_peer_query_position (auparse->sinkpad, GST_FORMAT_BYTES, &pos)) { GST_DEBUG_OBJECT (auparse, "failed to query upstream position"); break; } GST_OBJECT_LOCK (auparse); pos -= auparse->offset; GST_OBJECT_UNLOCK (auparse); ret = gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos, format, &val); if (ret) { gst_query_set_position (query, format, val); } break; } case GST_QUERY_SEEKING:{ GstFormat format; gst_query_parse_seeking (query, &format, NULL, NULL, NULL); /* FIXME: query duration in 'format' gst_query_set_seeking (query, format, TRUE, 0, duration); */ gst_query_set_seeking (query, format, TRUE, 0, GST_CLOCK_TIME_NONE); ret = TRUE; break; } default: ret = gst_pad_query_default (pad, parent, query); break; } return ret; } static gboolean gst_au_parse_handle_seek (GstAuParse * auparse, GstEvent * event) { GstSeekType start_type, stop_type; GstSeekFlags flags; GstFormat format; gdouble rate; gint64 start, stop; gboolean res; gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start, &stop_type, &stop); if (format != GST_FORMAT_TIME) { GST_DEBUG_OBJECT (auparse, "only support seeks in TIME format"); return FALSE; } res = gst_au_parse_src_convert (auparse, GST_FORMAT_TIME, start, GST_FORMAT_BYTES, &start); if (stop > 0) { res = gst_au_parse_src_convert (auparse, GST_FORMAT_TIME, stop, GST_FORMAT_BYTES, &stop); } GST_INFO_OBJECT (auparse, "seeking: %" G_GINT64_FORMAT " ... %" G_GINT64_FORMAT, start, stop); event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, start_type, start, stop_type, stop); res = gst_pad_push_event (auparse->sinkpad, event); return res; } static gboolean gst_au_parse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstAuParse *auparse; gboolean ret = TRUE; auparse = GST_AU_PARSE (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { /* discard, we'll come up with proper src caps */ gst_event_unref (event); break; } case GST_EVENT_SEGMENT: { gint64 start, stop, offset = 0; GstSegment segment; GstEvent *new_event = NULL; /* some debug output */ gst_event_copy_segment (event, &segment); GST_DEBUG_OBJECT (auparse, "received newsegment %" GST_SEGMENT_FORMAT, &segment); start = segment.start; stop = segment.stop; if (auparse->sample_size > 0) { if (start > 0) { offset = start; start -= auparse->offset; start = MAX (start, 0); } if (stop > 0) { stop -= auparse->offset; stop = MAX (stop, 0); } gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, start, GST_FORMAT_TIME, &start); gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, stop, GST_FORMAT_TIME, &stop); } GST_INFO_OBJECT (auparse, "new segment: %" GST_TIME_FORMAT " ... %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop)); gst_segment_init (&segment, GST_FORMAT_TIME); segment.start = segment.time = start; segment.stop = stop; new_event = gst_event_new_segment (&segment); ret = gst_pad_push_event (auparse->srcpad, new_event); auparse->buffer_offset = offset; gst_event_unref (event); break; } case GST_EVENT_EOS: if (!auparse->srcpad) { GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, ("No valid input found before end of stream"), (NULL)); } /* fall-through */ default: ret = gst_pad_event_default (pad, parent, event); break; } return ret; } static gboolean gst_au_parse_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstAuParse *auparse; gboolean ret; auparse = GST_AU_PARSE (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: ret = gst_au_parse_handle_seek (auparse, event); break; default: ret = gst_pad_event_default (pad, parent, event); break; } return ret; } static GstStateChangeReturn gst_au_parse_change_state (GstElement * element, GstStateChange transition) { GstAuParse *auparse = GST_AU_PARSE (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) return ret; switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_au_parse_reset (auparse); default: break; } return ret; } static gboolean plugin_init (GstPlugin * plugin) { if (!gst_element_register (plugin, "auparse", GST_RANK_SECONDARY, GST_TYPE_AU_PARSE)) { return FALSE; } return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, auparse, "parses au streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)