/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * 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. */ /* Element-Checklist-Version: 5 */ /* 2001/04/03 - Updated parseau to use caps nego * Zaheer Abbas Merali */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstauparse.h" #include /* elementfactory information */ static GstElementDetails gst_auparse_details = GST_ELEMENT_DETAILS (".au parser", "Codec/Demuxer/Audio", "Parse an .au file into raw audio", "Erik Walthinsen "); static GstStaticPadTemplate gst_auparse_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-au") ); static GstStaticPadTemplate gst_auparse_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_SOMETIMES, /* FIXME: spider */ GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; " /* we don't use GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS because of min buffer-frames which is 1, not 0 */ "audio/x-raw-float, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], " "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " "width = (int) { 32, 64 }, " "buffer-frames = (int) [ 0, MAX]" "; " "audio/x-alaw, " "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]" "; " "audio/x-mulaw, " "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]" "; " /* Nothing to decode those ADPCM streams for now */ "audio/x-adpcm, " "layout = (string) { g721, g722, g723_3, g723_5 }") ); enum { ARG_0 /* FILL ME */ }; static void gst_auparse_base_init (gpointer g_class); static void gst_auparse_class_init (GstAuParseClass * klass); static void gst_auparse_init (GstAuParse * auparse); static GstFlowReturn gst_auparse_chain (GstPad * pad, GstBuffer * buf); static GstStateChangeReturn gst_auparse_change_state (GstElement * element, GstStateChange transition); static GstElementClass *parent_class = NULL; /*static guint gst_auparse_signals[LAST_SIGNAL] = { 0 }; */ GType gst_auparse_get_type (void) { static GType auparse_type = 0; if (!auparse_type) { static const GTypeInfo auparse_info = { sizeof (GstAuParseClass), gst_auparse_base_init, NULL, (GClassInitFunc) gst_auparse_class_init, NULL, NULL, sizeof (GstAuParse), 0, (GInstanceInitFunc) gst_auparse_init, }; auparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAuParse", &auparse_info, 0); } return auparse_type; } static void gst_auparse_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_auparse_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_auparse_src_template)); gst_element_class_set_details (element_class, &gst_auparse_details); } static void gst_auparse_class_init (GstAuParseClass * klass) { GstElementClass *gstelement_class; gstelement_class = (GstElementClass *) klass; parent_class = g_type_class_ref (GST_TYPE_ELEMENT); gstelement_class->change_state = gst_auparse_change_state; } static void gst_auparse_init (GstAuParse * auparse) { auparse->sinkpad = gst_pad_new_from_template (gst_static_pad_template_get (&gst_auparse_sink_template), "sink"); gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad); gst_pad_set_chain_function (auparse->sinkpad, gst_auparse_chain); auparse->srcpad = gst_pad_new_from_template (gst_static_pad_template_get (&gst_auparse_src_template), "src"); gst_pad_use_fixed_caps (auparse->srcpad); gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad); auparse->offset = 0; auparse->size = 0; auparse->encoding = 0; auparse->frequency = 0; auparse->channels = 0; } static GstFlowReturn gst_auparse_chain (GstPad * pad, GstBuffer * buf) { GstFlowReturn ret; GstAuParse *auparse; guchar *data; glong size; GstCaps *tempcaps; gint law = 0, depth = 0, ieee = 0; gchar layout[7]; layout[0] = 0; auparse = GST_AUPARSE (gst_pad_get_parent (pad)); GST_DEBUG ("gst_auparse_chain: got buffer in '%s'", gst_element_get_name (GST_ELEMENT (auparse))); data = GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); /* if we haven't seen any data yet... */ if (auparse->size == 0) { GstBuffer *newbuf; guint32 *head = (guint32 *) data; /* normal format is big endian (au is a Sparc format) */ if (GST_READ_UINT32_BE (head) == 0x2e736e64) { /* ".snd" */ head++; auparse->le = 0; auparse->offset = GST_READ_UINT32_BE (head); head++; /* Do not trust size, could be set to -1 : unknown */ auparse->size = GST_READ_UINT32_BE (head); head++; auparse->encoding = GST_READ_UINT32_BE (head); head++; auparse->frequency = GST_READ_UINT32_BE (head); head++; auparse->channels = GST_READ_UINT32_BE (head); head++; /* and of course, someone had to invent a little endian * version. Used by DEC systems. */ } else if (GST_READ_UINT32_LE (head) == 0x0064732E) { /* other source say it is "dns." */ head++; auparse->le = 1; auparse->offset = GST_READ_UINT32_LE (head); head++; /* Do not trust size, could be set to -1 : unknown */ auparse->size = GST_READ_UINT32_LE (head); head++; auparse->encoding = GST_READ_UINT32_LE (head); head++; auparse->frequency = GST_READ_UINT32_LE (head); head++; auparse->channels = GST_READ_UINT32_LE (head); head++; } else { GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL)); gst_buffer_unref (buf); g_object_unref (auparse); return GST_FLOW_ERROR; } GST_DEBUG ("offset %ld, size %ld, encoding %ld, frequency %ld, channels %ld\n", auparse->offset, auparse->size, auparse->encoding, auparse->frequency, 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; depth = 8; break; case 27: /* 8-bit ISDN A-law G.711 */ law = 2; depth = 8; break; case 2: /* 8-bit linear PCM */ depth = 8; break; case 3: /* 16-bit linear PCM */ depth = 16; break; case 4: /* 24-bit linear PCM */ depth = 24; break; case 5: /* 32-bit linear PCM */ depth = 32; break; case 6: /* 32-bit IEEE floating point */ ieee = 1; depth = 32; break; case 7: /* 64-bit IEEE floating point */ ieee = 1; depth = 64; 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: GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), (NULL)); gst_buffer_unref (buf); g_object_unref (auparse); return GST_FLOW_ERROR; } if (law) { tempcaps = gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw", "rate", G_TYPE_INT, auparse->frequency, "channels", G_TYPE_INT, auparse->channels, NULL); } else if (ieee) { tempcaps = gst_caps_new_simple ("audio/x-raw-float", "rate", G_TYPE_INT, auparse->frequency, "channels", G_TYPE_INT, auparse->channels, "endianness", G_TYPE_INT, auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, "width", G_TYPE_INT, depth, "buffer-frames", G_TYPE_INT, 0, NULL); } else if (layout[0]) { tempcaps = gst_caps_new_simple ("audio/x-adpcm", "layout", G_TYPE_STRING, layout, NULL); } else { tempcaps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, auparse->frequency, "channels", G_TYPE_INT, auparse->channels, "endianness", G_TYPE_INT, auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, "depth", G_TYPE_INT, depth, "width", G_TYPE_INT, depth, "signed", G_TYPE_BOOLEAN, TRUE, NULL); } gst_pad_set_active (auparse->srcpad, TRUE); gst_pad_set_caps (auparse->srcpad, tempcaps); if ((ret = gst_pad_alloc_buffer (auparse->srcpad, GST_BUFFER_OFFSET_NONE, size - (auparse->offset), GST_PAD_CAPS (auparse->srcpad), &newbuf)) != GST_FLOW_OK) { gst_buffer_unref (buf); g_object_unref (auparse); return ret; } ret = GST_FLOW_OK; memcpy (GST_BUFFER_DATA (newbuf), data + (auparse->offset), size - (auparse->offset)); GST_BUFFER_SIZE (newbuf) = size - (auparse->offset); GstEvent *event; event = NULL; event = gst_event_new_newsegment (FALSE, 1.0, GST_FORMAT_DEFAULT, 0, GST_CLOCK_TIME_NONE, 0); gst_pad_push_event (auparse->srcpad, event); gst_buffer_unref (buf); g_object_unref (auparse); return gst_pad_push (auparse->srcpad, newbuf); } g_object_unref (auparse); return gst_pad_push (auparse->srcpad, buf); } static GstStateChangeReturn gst_auparse_change_state (GstElement * element, GstStateChange transition) { GstAuParse *auparse = GST_AUPARSE (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; if (parent_class->change_state) ret = parent_class->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_READY_TO_NULL: auparse->offset = 0; auparse->size = 0; auparse->encoding = 0; auparse->frequency = 0; auparse->channels = 0; default: break; } return ret; } static gboolean plugin_init (GstPlugin * plugin) { if (!gst_element_register (plugin, "auparse", GST_RANK_SECONDARY, GST_TYPE_AUPARSE)) { 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)