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gstreamer/ext/ogg/gstogmparse.c
Stéphane Loeuillet 627ff3b293 fix ogm[audio/video]parse plugin registration (riff won't load if bytestream is already loaded) 
Original commit message from CVS:
fix ogm[audio/video]parse plugin registration
(riff won't load if bytestream is already loaded)
2004-12-26 19:00:21 +00:00

554 lines
17 KiB
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/* GStreamer
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* gstogmparse.c: OGM stream header parsing (and data passthrough)
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/riff/riff-media.h>
GST_DEBUG_CATEGORY_STATIC (gst_ogm_parse_debug);
#define GST_CAT_DEFAULT gst_ogm_parse_debug
#define GST_TYPE_OGM_VIDEO_PARSE (gst_ogm_video_parse_get_type())
#define GST_IS_OGM_VIDEO_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_OGM_VIDEO_PARSE))
#define GST_TYPE_OGM_AUDIO_PARSE (gst_ogm_audio_parse_get_type())
#define GST_IS_OGM_AUDIO_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_OGM_AUDIO_PARSE))
#define GST_TYPE_OGM_PARSE (gst_ogm_parse_get_type())
#define GST_OGM_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_OGM_PARSE, GstOgmParse))
#define GST_OGM_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_OGM_PARSE, GstOgmParse))
#define GST_IS_OGM_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_OGM_PARSE))
#define GST_IS_OGM_PARSE_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_OGM_PARSE))
#define GST_OGM_PARSE_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_OGM_PARSE, GstOgmParseClass))
typedef struct _stream_header_video
{
gint32 width;
gint32 height;
} stream_header_video;
typedef struct _stream_header_audio
{
gint16 channels;
gint16 blockalign;
gint32 avgbytespersec;
} stream_header_audio;
typedef struct _stream_header
{
gchar streamtype[8];
gchar subtype[4];
/* size of the structure */
gint32 size;
/* in reference time */
gint64 time_unit;
gint64 samples_per_unit;
/* in media time */
gint32 default_len;
gint32 buffersize;
gint32 bits_per_sample;
union
{
stream_header_video video;
stream_header_audio audio;
} s;
} stream_header;
typedef struct _GstOgmParse
{
GstElement element;
/* pads */
GstPad *srcpad, *sinkpad;
GstPadTemplate *srcpadtempl;
/* audio or video */
stream_header hdr;
/* expected next granulepos (used for timestamp guessing) */
guint64 next_granulepos;
} GstOgmParse;
typedef struct _GstOgmParseClass
{
GstElementClass parent_class;
} GstOgmParseClass;
static GstStaticPadTemplate ogm_video_parse_sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-ogm-video"));
static GstStaticPadTemplate ogm_audio_parse_sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-ogm-audio"));
static GstPadTemplate *video_src_templ, *audio_src_templ;
static GType gst_ogm_audio_parse_get_type (void);
static GType gst_ogm_video_parse_get_type (void);
static GType gst_ogm_parse_get_type (void);
static void gst_ogm_audio_parse_base_init (GstOgmParseClass * klass);
static void gst_ogm_video_parse_base_init (GstOgmParseClass * klass);
static void gst_ogm_parse_class_init (GstOgmParseClass * klass);
static void gst_ogm_parse_init (GstOgmParse * ogm);
static void gst_ogm_video_parse_init (GstOgmParse * ogm);
static void gst_ogm_audio_parse_init (GstOgmParse * ogm);
static const GstFormat *gst_ogm_parse_get_sink_formats (GstPad * pad);
static gboolean gst_ogm_parse_sink_convert (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static void gst_ogm_parse_chain (GstPad * pad, GstData * data);
static GstElementStateReturn gst_ogm_parse_change_state (GstElement * element);
GstElementClass *parent_class = NULL;
static GType
gst_ogm_parse_get_type (void)
{
static GType ogm_parse_type = 0;
if (!ogm_parse_type) {
static const GTypeInfo ogm_parse_info = {
sizeof (GstOgmParseClass),
NULL,
NULL,
(GClassInitFunc) gst_ogm_parse_class_init,
NULL,
NULL,
sizeof (GstOgmParse),
0,
(GInstanceInitFunc) gst_ogm_parse_init,
};
ogm_parse_type =
g_type_register_static (GST_TYPE_ELEMENT,
"GstOgmParse", &ogm_parse_info, 0);
}
return ogm_parse_type;
}
static GType
gst_ogm_audio_parse_get_type (void)
{
static GType ogm_audio_parse_type = 0;
if (!ogm_audio_parse_type) {
static const GTypeInfo ogm_audio_parse_info = {
sizeof (GstOgmParseClass),
(GBaseInitFunc) gst_ogm_audio_parse_base_init,
NULL,
NULL,
NULL,
NULL,
sizeof (GstOgmParse),
0,
(GInstanceInitFunc) gst_ogm_audio_parse_init,
};
ogm_audio_parse_type =
g_type_register_static (GST_TYPE_OGM_PARSE,
"GstOgmAudioParse", &ogm_audio_parse_info, 0);
}
return ogm_audio_parse_type;
}
GType
gst_ogm_video_parse_get_type (void)
{
static GType ogm_video_parse_type = 0;
if (!ogm_video_parse_type) {
static const GTypeInfo ogm_video_parse_info = {
sizeof (GstOgmParseClass),
(GBaseInitFunc) gst_ogm_video_parse_base_init,
NULL,
NULL,
NULL,
NULL,
sizeof (GstOgmParse),
0,
(GInstanceInitFunc) gst_ogm_video_parse_init,
};
ogm_video_parse_type =
g_type_register_static (GST_TYPE_OGM_PARSE,
"GstOgmVideoParse", &ogm_video_parse_info, 0);
}
return ogm_video_parse_type;
}
static void
gst_ogm_audio_parse_base_init (GstOgmParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_ogm_audio_parse_details =
GST_ELEMENT_DETAILS ("OGM audio stream parser",
"Codec/Decoder/Audio",
"parse an OGM audio header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GstCaps *caps = gst_riff_create_audio_template_caps ();
gst_element_class_set_details (element_class, &gst_ogm_audio_parse_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&ogm_audio_parse_sink_template_factory));
audio_src_templ = gst_pad_template_new ("src",
GST_PAD_SRC, GST_PAD_SOMETIMES, caps);
gst_element_class_add_pad_template (element_class, audio_src_templ);
}
static void
gst_ogm_video_parse_base_init (GstOgmParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_ogm_video_parse_details =
GST_ELEMENT_DETAILS ("OGM video stream parser",
"Codec/Decoder/Video",
"parse an OGM video header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GstCaps *caps = gst_riff_create_video_template_caps ();
gst_element_class_set_details (element_class, &gst_ogm_video_parse_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&ogm_video_parse_sink_template_factory));
video_src_templ = gst_pad_template_new ("src",
GST_PAD_SRC, GST_PAD_SOMETIMES, caps);
gst_element_class_add_pad_template (element_class, video_src_templ);
}
static void
gst_ogm_parse_class_init (GstOgmParseClass * klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gstelement_class->change_state = gst_ogm_parse_change_state;
}
static void
gst_ogm_parse_init (GstOgmParse * ogm)
{
/* initalize */
memset (&ogm->hdr, 0, sizeof (ogm->hdr));
ogm->next_granulepos = 0;
}
static void
gst_ogm_audio_parse_init (GstOgmParse * ogm)
{
GstPadTemplate *templ;
/* create the pads */
templ = gst_static_pad_template_get (&ogm_audio_parse_sink_template_factory);
ogm->sinkpad = gst_pad_new_from_template (templ, "sink");
gst_pad_set_convert_function (ogm->sinkpad, gst_ogm_parse_sink_convert);
gst_pad_set_formats_function (ogm->sinkpad, gst_ogm_parse_get_sink_formats);
gst_pad_set_chain_function (ogm->sinkpad, gst_ogm_parse_chain);
gst_element_add_pad (GST_ELEMENT (ogm), ogm->sinkpad);
#if 0
ogm->srcpad = gst_pad_new_from_template (audio_src_templ, "src");
gst_pad_use_explicit_caps (ogm->srcpad);
gst_element_add_pad (GST_ELEMENT (ogm), ogm->srcpad);
#endif
ogm->srcpadtempl = audio_src_templ;
}
static void
gst_ogm_video_parse_init (GstOgmParse * ogm)
{
GstPadTemplate *templ;
/* create the pads */
templ = gst_static_pad_template_get (&ogm_video_parse_sink_template_factory);
ogm->sinkpad = gst_pad_new_from_template (templ, "sink");
gst_pad_set_convert_function (ogm->sinkpad, gst_ogm_parse_sink_convert);
gst_pad_set_formats_function (ogm->sinkpad, gst_ogm_parse_get_sink_formats);
gst_pad_set_chain_function (ogm->sinkpad, gst_ogm_parse_chain);
gst_element_add_pad (GST_ELEMENT (ogm), ogm->sinkpad);
#if 0
ogm->srcpad = gst_pad_new_from_template (video_src_templ, "src");
gst_pad_use_explicit_caps (ogm->srcpad);
gst_element_add_pad (GST_ELEMENT (ogm), ogm->srcpad);
#endif
ogm->srcpadtempl = video_src_templ;
}
static const GstFormat *
gst_ogm_parse_get_sink_formats (GstPad * pad)
{
static GstFormat formats[] = {
GST_FORMAT_DEFAULT,
GST_FORMAT_TIME,
0
};
return formats;
}
static gboolean
gst_ogm_parse_sink_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
GstOgmParse *ogm = GST_OGM_PARSE (gst_pad_get_parent (pad));
switch (src_format) {
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_TIME:
switch (ogm->hdr.streamtype[0]) {
case 'a':
*dest_value = GST_SECOND * src_value / ogm->hdr.samples_per_unit;
res = TRUE;
break;
case 'v':
*dest_value = (GST_SECOND / 10000000) *
ogm->hdr.time_unit * src_value;
res = TRUE;
break;
default:
break;
}
break;
default:
break;
}
break;
default:
break;
}
return res;
}
static void
gst_ogm_parse_chain (GstPad * pad, GstData * dat)
{
GstOgmParse *ogm = GST_OGM_PARSE (gst_pad_get_parent (pad));
GstBuffer *buf = GST_BUFFER (dat);
guint8 *data = GST_BUFFER_DATA (buf);
guint size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (ogm, "New packet with packet start code 0x%02x", data[0]);
switch (data[0]) {
case 0x01:{
GstCaps *caps = NULL;
/* stream header */
if (size < sizeof (stream_header) + 1) {
GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
("Buffer too small"), (NULL));
break;
}
if (!memcmp (&data[1], "video\000\000\000", 8)) {
ogm->hdr.s.video.width = GST_READ_UINT32_LE (&data[45]);
ogm->hdr.s.video.height = GST_READ_UINT32_LE (&data[49]);
} else if (!memcmp (&data[1], "audio\000\000\000", 8)) {
ogm->hdr.s.audio.channels = GST_READ_UINT32_LE (&data[45]);
ogm->hdr.s.audio.blockalign = GST_READ_UINT32_LE (&data[47]);
ogm->hdr.s.audio.avgbytespersec = GST_READ_UINT32_LE (&data[49]);
} else {
GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
("Unknown stream type"), (NULL));
break;
}
memcpy (ogm->hdr.streamtype, &data[1], 8);
memcpy (ogm->hdr.subtype, &data[9], 4);
ogm->hdr.size = GST_READ_UINT32_LE (&data[13]);
ogm->hdr.time_unit = GST_READ_UINT64_LE (&data[17]);
ogm->hdr.samples_per_unit = GST_READ_UINT64_LE (&data[25]);
ogm->hdr.default_len = GST_READ_UINT32_LE (&data[33]);
ogm->hdr.buffersize = GST_READ_UINT32_LE (&data[37]);
ogm->hdr.bits_per_sample = GST_READ_UINT32_LE (&data[41]);
switch (ogm->hdr.streamtype[0]) {
case 'a':{
guint codec_id;
if (!sscanf (ogm->hdr.subtype, "%04x", &codec_id)) {
caps = NULL;
break;
}
caps = gst_riff_create_audio_caps (codec_id, NULL, NULL, NULL);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, ogm->hdr.s.audio.channels,
"rate", G_TYPE_INT, ogm->hdr.samples_per_unit, NULL);
GST_LOG_OBJECT (ogm, "Type: %s, subtype: 0x%04x, "
"channels: %d, samplerate: %d, blockalign: %d, bps: %d",
ogm->hdr.streamtype, codec_id, ogm->hdr.s.audio.channels,
ogm->hdr.samples_per_unit,
ogm->hdr.s.audio.blockalign, ogm->hdr.s.audio.avgbytespersec);
break;
}
case 'v':{
guint32 fcc;
fcc = GST_MAKE_FOURCC (ogm->hdr.subtype[0],
ogm->hdr.subtype[1], ogm->hdr.subtype[2], ogm->hdr.subtype[3]);
GST_LOG_OBJECT (ogm, "Type: %s, subtype: " GST_FOURCC_FORMAT
", size: %dx%d, timeunit: %" G_GINT64_FORMAT
" (fps: %lf), s/u: %" G_GINT64_FORMAT ", "
"def.len: %d, bufsize: %d, bps: %d",
ogm->hdr.streamtype, GST_FOURCC_ARGS (fcc),
ogm->hdr.s.video.width, ogm->hdr.s.video.height,
ogm->hdr.time_unit, 10000000. / ogm->hdr.time_unit,
ogm->hdr.samples_per_unit, ogm->hdr.default_len,
ogm->hdr.buffersize, ogm->hdr.bits_per_sample);
caps = gst_riff_create_video_caps (fcc, NULL, NULL, NULL);
gst_caps_set_simple (caps,
"width", G_TYPE_INT, ogm->hdr.s.video.width,
"height", G_TYPE_INT, ogm->hdr.s.video.height,
"framerate", G_TYPE_DOUBLE, 10000000. / ogm->hdr.time_unit, NULL);
break;
}
default:
g_assert_not_reached ();
}
ogm->srcpad = gst_pad_new_from_template (ogm->srcpadtempl, "src");
gst_pad_use_explicit_caps (ogm->srcpad);
if (!gst_pad_set_explicit_caps (ogm->srcpad, caps)) {
GST_ELEMENT_ERROR (ogm, CORE, NEGOTIATION, (NULL), (NULL));
//gst_object_unref (GST_OBJECT (ogm->srcpad));
ogm->srcpad = NULL;
break;
}
gst_element_add_pad (GST_ELEMENT (ogm), ogm->srcpad);
break;
}
case 0x03:
/* comment - unused */
break;
default:
if ((data[0] & 0x01) == 0) {
/* data - push on */
guint len = ((data[0] & 0xc0) >> 6) | ((data[0] & 0x02) << 1);
guint xsize = 0, n;
GstBuffer *sbuf;
gboolean keyframe = (data[0] & 0x08) >> 3;
if (size < len + 1) {
GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
("Buffer too small"), (NULL));
break;
}
for (n = len; n > 0; n--) {
xsize = (xsize << 8) | data[n];
}
GST_DEBUG_OBJECT (ogm,
"[0x%02x] samples: %d, hdrbytes: %d, datasize: %d",
data[0], xsize, len, size - len - 1);
sbuf = gst_buffer_create_sub (buf, len + 1, size - len - 1);
if (GST_BUFFER_OFFSET_END_IS_VALID (buf)) {
ogm->next_granulepos = GST_BUFFER_OFFSET_END (buf);
}
switch (ogm->hdr.streamtype[0]) {
case 'v':
if (keyframe)
GST_BUFFER_FLAG_SET (sbuf, GST_BUFFER_KEY_UNIT);
GST_BUFFER_TIMESTAMP (sbuf) = (GST_SECOND / 10000000) *
ogm->next_granulepos * ogm->hdr.time_unit;
GST_BUFFER_DURATION (sbuf) = (GST_SECOND / 10000000) *
ogm->hdr.time_unit;
ogm->next_granulepos++;
break;
case 'a':
GST_BUFFER_TIMESTAMP (sbuf) = GST_SECOND *
ogm->next_granulepos / ogm->hdr.samples_per_unit;
GST_BUFFER_DURATION (sbuf) = GST_SECOND * xsize /
ogm->hdr.samples_per_unit;
ogm->next_granulepos += xsize;
break;
default:
g_assert_not_reached ();
}
gst_pad_push (ogm->srcpad, GST_DATA (sbuf));
} else {
GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
("Wrong packet startcode 0x%02x", data[0]), (NULL));
}
break;
}
gst_buffer_unref (buf);
}
static GstElementStateReturn
gst_ogm_parse_change_state (GstElement * element)
{
GstOgmParse *ogm = GST_OGM_PARSE (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_PAUSED_TO_READY:
if (ogm->srcpad) {
gst_element_remove_pad (element, ogm->srcpad);
ogm->srcpad = NULL;
}
memset (&ogm->hdr, 0, sizeof (ogm->hdr));
ogm->next_granulepos = 0;
break;
default:
break;
}
return parent_class->change_state (element);
}
gboolean
gst_ogm_parse_plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (gst_ogm_parse_debug, "ogmparse", 0, "ogm parser");
return gst_element_register (plugin, "ogmaudioparse", GST_RANK_PRIMARY,
GST_TYPE_OGM_AUDIO_PARSE)
&& gst_element_register (plugin, "ogmvideoparse", GST_RANK_PRIMARY,
GST_TYPE_OGM_VIDEO_PARSE);
}
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