/* GStreamer * Copyright (C) 2006 Wim Taymans * * gstoggaviparse.c: ogg avi stream parser * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /* * Ogg in AVI is mostly done for vorbis audio. In the codec_data we receive the * first 3 packets of the raw vorbis data. On the sinkpad we receive full-blown Ogg * pages. * Before extracting the packets out of the ogg pages, we push the raw vorbis * header packets to the decoder. * We don't use the incomming timestamps but use the ganulepos on the ogg pages * directly. * This parser only does ogg/vorbis for now. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include "gstogg.h" GST_DEBUG_CATEGORY_STATIC (gst_ogg_avi_parse_debug); #define GST_CAT_DEFAULT gst_ogg_avi_parse_debug #define GST_TYPE_OGG_AVI_PARSE (gst_ogg_avi_parse_get_type()) #define GST_OGG_AVI_PARSE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OGG_AVI_PARSE, GstOggAviParse)) #define GST_OGG_AVI_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OGG_AVI_PARSE, GstOggAviParse)) #define GST_IS_OGG_AVI_PARSE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OGG_AVI_PARSE)) #define GST_IS_OGG_AVI_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OGG_AVI_PARSE)) static GType gst_ogg_avi_parse_get_type (void); typedef struct _GstOggAviParse GstOggAviParse; typedef struct _GstOggAviParseClass GstOggAviParseClass; struct _GstOggAviParse { GstElement element; GstPad *sinkpad; GstPad *srcpad; gboolean discont; gint serial; ogg_sync_state sync; ogg_stream_state stream; }; struct _GstOggAviParseClass { GstElementClass parent_class; }; static void gst_ogg_avi_parse_base_init (gpointer g_class); static void gst_ogg_avi_parse_class_init (GstOggAviParseClass * klass); static void gst_ogg_avi_parse_init (GstOggAviParse * ogg); static GstElementClass *parent_class = NULL; static GType gst_ogg_avi_parse_get_type (void) { static GType ogg_avi_parse_type = 0; if (!ogg_avi_parse_type) { static const GTypeInfo ogg_avi_parse_info = { sizeof (GstOggAviParseClass), gst_ogg_avi_parse_base_init, NULL, (GClassInitFunc) gst_ogg_avi_parse_class_init, NULL, NULL, sizeof (GstOggAviParse), 0, (GInstanceInitFunc) gst_ogg_avi_parse_init, }; ogg_avi_parse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstOggAviParse", &ogg_avi_parse_info, 0); } return ogg_avi_parse_type; } enum { PROP_0 }; static GstStaticPadTemplate ogg_avi_parse_src_template_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-vorbis") ); static GstStaticPadTemplate ogg_avi_parse_sink_template_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-ogg-avi") ); static void gst_ogg_avi_parse_finalize (GObject * object); static GstStateChangeReturn gst_ogg_avi_parse_change_state (GstElement * element, GstStateChange transition); static gboolean gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static gboolean gst_ogg_avi_parse_setcaps (GstPad * pad, GstCaps * caps); static void gst_ogg_avi_parse_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_static_metadata (element_class, "Ogg AVI parser", "Codec/Parser", "parse an ogg avi stream into pages (info about ogg: http://xiph.org)", "Wim Taymans "); gst_element_class_add_static_pad_template (element_class, &ogg_avi_parse_sink_template_factory); gst_element_class_add_static_pad_template (element_class, &ogg_avi_parse_src_template_factory); } static void gst_ogg_avi_parse_class_init (GstOggAviParseClass * klass) { GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GObjectClass *gobject_class = G_OBJECT_CLASS (klass); parent_class = g_type_class_peek_parent (klass); gstelement_class->change_state = gst_ogg_avi_parse_change_state; gobject_class->finalize = gst_ogg_avi_parse_finalize; } static void gst_ogg_avi_parse_init (GstOggAviParse * ogg) { /* create the sink and source pads */ ogg->sinkpad = gst_pad_new_from_static_template (&ogg_avi_parse_sink_template_factory, "sink"); gst_pad_set_event_function (ogg->sinkpad, gst_ogg_avi_parse_event); gst_pad_set_chain_function (ogg->sinkpad, gst_ogg_avi_parse_chain); gst_element_add_pad (GST_ELEMENT (ogg), ogg->sinkpad); ogg->srcpad = gst_pad_new_from_static_template (&ogg_avi_parse_src_template_factory, "src"); gst_pad_use_fixed_caps (ogg->srcpad); gst_element_add_pad (GST_ELEMENT (ogg), ogg->srcpad); } static void gst_ogg_avi_parse_finalize (GObject * object) { GstOggAviParse *ogg = GST_OGG_AVI_PARSE (object); GST_LOG_OBJECT (ogg, "Disposing of object %p", ogg); ogg_sync_clear (&ogg->sync); ogg_stream_clear (&ogg->stream); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_ogg_avi_parse_setcaps (GstPad * pad, GstCaps * caps) { GstOggAviParse *ogg; GstStructure *structure; const GValue *codec_data; GstBuffer *buffer; GstMapInfo map; guint8 *ptr; gsize left; guint32 sizes[3]; GstCaps *outcaps; gint i, offs; ogg = GST_OGG_AVI_PARSE (GST_OBJECT_PARENT (pad)); structure = gst_caps_get_structure (caps, 0); /* take codec data */ codec_data = gst_structure_get_value (structure, "codec_data"); if (codec_data == NULL) goto no_data; /* only buffers are valid */ if (G_VALUE_TYPE (codec_data) != GST_TYPE_BUFFER) goto wrong_format; /* Now parse the data */ buffer = gst_value_get_buffer (codec_data); /* first 22 bytes are bits_per_sample, channel_mask, GUID * Then we get 3 LE guint32 with the 3 header sizes * then we get the bytes of the 3 headers. */ gst_buffer_map (buffer, &map, GST_MAP_READ); ptr = map.data; left = map.size; GST_LOG_OBJECT (ogg, "configuring codec_data of size %" G_GSIZE_FORMAT, left); /* skip headers */ ptr += 22; left -= 22; /* we need at least 12 bytes for the packet sizes of the 3 headers */ if (left < 12) goto buffer_too_small; /* read sizes of the 3 headers */ sizes[0] = GST_READ_UINT32_LE (ptr); sizes[1] = GST_READ_UINT32_LE (ptr + 4); sizes[2] = GST_READ_UINT32_LE (ptr + 8); GST_DEBUG_OBJECT (ogg, "header sizes: %u %u %u", sizes[0], sizes[1], sizes[2]); left -= 12; /* and we need at least enough data for all the headers */ if (left < sizes[0] + sizes[1] + sizes[2]) goto buffer_too_small; /* set caps */ outcaps = gst_caps_new_empty_simple ("audio/x-vorbis"); gst_pad_set_caps (ogg->srcpad, outcaps); gst_caps_unref (outcaps); /* copy header data */ offs = 34; for (i = 0; i < 3; i++) { GstBuffer *out; /* now output the raw vorbis header packets */ out = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offs, sizes[i]); gst_pad_push (ogg->srcpad, out); offs += sizes[i]; } gst_buffer_unmap (buffer, &map); return TRUE; /* ERRORS */ no_data: { GST_DEBUG_OBJECT (ogg, "no codec_data found in caps"); return FALSE; } wrong_format: { GST_DEBUG_OBJECT (ogg, "codec_data is not a buffer"); return FALSE; } buffer_too_small: { GST_DEBUG_OBJECT (ogg, "codec_data is too small"); gst_buffer_unmap (buffer, &map); return FALSE; } } static gboolean gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstOggAviParse *ogg; gboolean ret; ogg = GST_OGG_AVI_PARSE (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); ret = gst_ogg_avi_parse_setcaps (pad, caps); gst_event_unref (event); break; } case GST_EVENT_FLUSH_START: ret = gst_pad_push_event (ogg->srcpad, event); break; case GST_EVENT_FLUSH_STOP: ogg_sync_reset (&ogg->sync); ogg_stream_reset (&ogg->stream); ogg->discont = TRUE; ret = gst_pad_push_event (ogg->srcpad, event); break; default: ret = gst_pad_push_event (ogg->srcpad, event); break; } return ret; } static GstFlowReturn gst_ogg_avi_parse_push_packet (GstOggAviParse * ogg, ogg_packet * packet) { GstBuffer *buffer; GstFlowReturn result; /* allocate space for header and body */ buffer = gst_buffer_new_and_alloc (packet->bytes); gst_buffer_fill (buffer, 0, packet->packet, packet->bytes); GST_LOG_OBJECT (ogg, "created buffer %p from page", buffer); GST_BUFFER_OFFSET_END (buffer) = packet->granulepos; if (ogg->discont) { GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); ogg->discont = FALSE; } result = gst_pad_push (ogg->srcpad, buffer); return result; } static GstFlowReturn gst_ogg_avi_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstFlowReturn result = GST_FLOW_OK; GstOggAviParse *ogg; guint size; gchar *oggbuf; gint ret = -1; ogg = GST_OGG_AVI_PARSE (parent); size = gst_buffer_get_size (buffer); GST_LOG_OBJECT (ogg, "Chain function received buffer of size %d", size); if (GST_BUFFER_IS_DISCONT (buffer)) { ogg_sync_reset (&ogg->sync); ogg->discont = TRUE; } /* write data to sync layer */ oggbuf = ogg_sync_buffer (&ogg->sync, size); gst_buffer_extract (buffer, 0, oggbuf, size); ogg_sync_wrote (&ogg->sync, size); gst_buffer_unref (buffer); /* try to get as many packets out of the stream as possible */ do { ogg_page page; /* try to swap out a page */ ret = ogg_sync_pageout (&ogg->sync, &page); if (ret == 0) { GST_DEBUG_OBJECT (ogg, "need more data"); break; } else if (ret == -1) { GST_DEBUG_OBJECT (ogg, "discont in pages"); ogg->discont = TRUE; } else { /* new unknown stream, init the ogg stream with the serial number of the * page. */ if (ogg->serial == -1) { ogg->serial = ogg_page_serialno (&page); ogg_stream_init (&ogg->stream, ogg->serial); } /* submit page */ if (ogg_stream_pagein (&ogg->stream, &page) != 0) { GST_WARNING_OBJECT (ogg, "ogg stream choked on page resetting stream"); ogg_sync_reset (&ogg->sync); ogg->discont = TRUE; continue; } /* try to get as many packets as possible out of the page */ do { ogg_packet packet; ret = ogg_stream_packetout (&ogg->stream, &packet); GST_LOG_OBJECT (ogg, "packetout gave %d", ret); switch (ret) { case 0: break; case -1: /* out of sync, We mark a DISCONT. */ ogg->discont = TRUE; break; case 1: result = gst_ogg_avi_parse_push_packet (ogg, &packet); if (result != GST_FLOW_OK) goto done; break; default: GST_WARNING_OBJECT (ogg, "invalid return value %d for ogg_stream_packetout, resetting stream", ret); break; } } while (ret != 0); } } while (ret != 0); done: return result; } static GstStateChangeReturn gst_ogg_avi_parse_change_state (GstElement * element, GstStateChange transition) { GstOggAviParse *ogg; GstStateChangeReturn result = GST_STATE_CHANGE_FAILURE; ogg = GST_OGG_AVI_PARSE (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: ogg_sync_init (&ogg->sync); break; case GST_STATE_CHANGE_READY_TO_PAUSED: ogg_sync_reset (&ogg->sync); ogg_stream_reset (&ogg->stream); ogg->serial = -1; ogg->discont = TRUE; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; default: break; } result = parent_class->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: break; case GST_STATE_CHANGE_READY_TO_NULL: ogg_sync_clear (&ogg->sync); break; default: break; } return result; } gboolean gst_ogg_avi_parse_plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (gst_ogg_avi_parse_debug, "oggaviparse", 0, "ogg avi parser"); return gst_element_register (plugin, "oggaviparse", GST_RANK_PRIMARY, GST_TYPE_OGG_AVI_PARSE); }