gstreamer/subprojects/gst-plugins-base/ext/ogg/gstoggaviparse.c

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/* 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., 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.
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* We don't use the incoming 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 "gstoggelements.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 GstElementClass *parent_class = NULL;
G_DEFINE_TYPE (GstOggAviParse, gst_ogg_avi_parse, GST_TYPE_ELEMENT);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (oggaviparse, "oggaviparse",
GST_RANK_PRIMARY, GST_TYPE_OGG_AVI_PARSE,
GST_DEBUG_CATEGORY_INIT (gst_ogg_avi_parse_debug, "oggaviparse", 0,
"ogg avi parser"));
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,
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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_class_init (GstOggAviParseClass * klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
gst_element_class_set_static_metadata (gstelement_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_static_pad_template (gstelement_class,
&ogg_avi_parse_sink_template_factory);
gst_element_class_add_static_pad_template (gstelement_class,
&ogg_avi_parse_src_template_factory);
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;
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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. */
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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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 */
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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 */
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out = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offs, sizes[i]);
gst_pad_push (ogg->srcpad, out);
offs += sizes[i];
}
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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");
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gst_buffer_unmap (buffer, &map);
return FALSE;
}
}
static gboolean
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gst_ogg_avi_parse_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstOggAviParse *ogg;
gboolean ret;
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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
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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;
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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;
}