--- /dev/null
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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 <gst/gst.h>
+#include <ogg/ogg.h>
+#include <string.h>
+
+#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, GstEvent * event);
+static GstFlowReturn gst_ogg_avi_parse_chain (GstPad * pad, 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_details_simple (element_class,
+ "Ogg AVI parser", "Codec/Parser",
+ "parse an ogg avi stream into pages (info about ogg: http://xiph.org)",
+ "Wim Taymans <wim@fluendo.com>");
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&ogg_avi_parse_sink_template_factory));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&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_setcaps_function (ogg->sinkpad, gst_ogg_avi_parse_setcaps);
+ 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;
+ guint8 *data;
+ guint size;
+ 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. */
+ data = GST_BUFFER_DATA (buffer);
+ size = GST_BUFFER_SIZE (buffer);
+
+ GST_LOG_OBJECT (ogg, "configuring codec_data of size %u", size);
+
+ /* skip headers */
+ data += 22;
+ size -= 22;
+
+ /* we need at least 12 bytes for the packet sizes of the 3 headers */
+ if (size < 12)
+ goto buffer_too_small;
+
+ /* read sizes of the 3 headers */
+ sizes[0] = GST_READ_UINT32_LE (data);
+ sizes[1] = GST_READ_UINT32_LE (data + 4);
+ sizes[2] = GST_READ_UINT32_LE (data + 8);
+
+ GST_DEBUG_OBJECT (ogg, "header sizes: %u %u %u", sizes[0], sizes[1],
+ sizes[2]);
+
+ size -= 12;
+
+ /* and we need at least enough data for all the headers */
+ if (size < sizes[0] + sizes[1] + sizes[2])
+ goto buffer_too_small;
+
+ /* set caps */
+ outcaps = gst_caps_new_simple ("audio/x-vorbis", NULL);
+ gst_pad_set_caps (ogg->srcpad, outcaps);
+
+ /* copy header data */
+ offs = 34;
+ for (i = 0; i < 3; i++) {
+ GstBuffer *out;
+
+ /* now output the raw vorbis header packets */
+ out = gst_buffer_create_sub (buffer, offs, sizes[i]);
+ gst_buffer_set_caps (out, outcaps);
+ gst_pad_push (ogg->srcpad, out);
+
+ offs += sizes[i];
+ }
+ gst_caps_unref (outcaps);
+
+ 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");
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_ogg_avi_parse_event (GstPad * pad, GstEvent * event)
+{
+ GstOggAviParse *ogg;
+ gboolean ret;
+
+ ogg = GST_OGG_AVI_PARSE (GST_OBJECT_PARENT (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ 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);
+ memcpy (GST_BUFFER_DATA (buffer), 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, GstBuffer * buffer)
+{
+ GstFlowReturn result = GST_FLOW_OK;
+ GstOggAviParse *ogg;
+ guint8 *data;
+ guint size;
+ gchar *oggbuf;
+ gint ret = -1;
+
+ ogg = GST_OGG_AVI_PARSE (GST_OBJECT_PARENT (pad));
+
+ data = GST_BUFFER_DATA (buffer);
+ size = GST_BUFFER_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);
+ memcpy (oggbuf, data, 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);
+}
projects / mafwsubrenderer / blobdiff