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823ceafca7
Original commit message from CVS: * ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_change_state): Allow for mp3 rate/channels changes. However, only very conservatively. Reason that we *have* to enable this is smiply because the mad find_sync() function is not good enough, it will regularly sync on random data as valid frames and therefore make us provide random caps as *final* caps of the stream. The best fix I could think of is to simply require several of the same stream changes in a row before we change caps. The actual testcase that works now is # * ext/ogg/Makefile.am: * ext/ogg/gstogg.c: (plugin_init): * ext/ogg/gstogmparse.c: OGM support (video only for now; I need an audio sample file). * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init), (gst_asf_demux_process_stream), (gst_asf_demux_video_caps), (gst_asf_demux_add_video_stream): WMV extradata. * gst/playback/gstplaybasebin.c: (unknown_type): Don't error out on single unknown-types after all. It's wrong. If we found type of video and audio but not of a subtitle stream, it will still error out (which is unwanted). Will find a better fix later on. * gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find), (ogmaudio_type_find), (plugin_init): OGM support.
485 lines
14 KiB
C
485 lines
14 KiB
C
/* GStreamer
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* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
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*
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* gstogmparse.c: OGM stream header parsing (and data passthrough)
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/riff/riff-media.h>
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GST_DEBUG_CATEGORY_STATIC (gst_ogm_parse_debug);
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#define GST_CAT_DEFAULT gst_ogm_parse_debug
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#define GST_TYPE_OGM_VIDEO_PARSE (gst_ogm_video_parse_get_type())
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#define GST_TYPE_OGM_AUDIO_PARSE (gst_ogm_audio_parse_get_type())
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#define GST_TYPE_OGM_PARSE (gst_ogm_parse_get_type())
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#define GST_OGM_PARSE(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_OGM_PARSE, GstOgmParse))
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#define GST_OGM_PARSE_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_OGM_PARSE, GstOgmParse))
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#define GST_IS_OGM_PARSE(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_OGM_PARSE))
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#define GST_IS_OGM_PARSE_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_OGM_PARSE))
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typedef struct _stream_header_video
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{
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gint32 width;
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gint32 height;
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} stream_header_video;
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typedef struct _stream_header_audio
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{
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gint16 channels;
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gint16 blockalign;
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gint32 avgbytespersec;
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} stream_header_audio;
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typedef struct _stream_header
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{
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gchar streamtype[8];
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gchar subtype[4];
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/* size of the structure */
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gint32 size;
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/* in reference time */
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gint64 time_unit;
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gint64 samples_per_unit;
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/* in media time */
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gint32 default_len;
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gint32 buffersize;
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gint32 bits_per_sample;
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union
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{
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stream_header_video video;
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stream_header_audio audio;
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} s;
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} stream_header;
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typedef struct _GstOgmParse
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{
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GstElement element;
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/* pads */
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GstPad *srcpad, *sinkpad;
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/* audio or video */
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stream_header hdr;
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} GstOgmParse;
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typedef struct _GstOgmParseClass
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{
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GstElementClass parent_class;
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} GstOgmParseClass;
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static GstStaticPadTemplate ogm_video_parse_sink_template_factory =
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GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-ogm-video"));
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static GstStaticPadTemplate ogm_audio_parse_sink_template_factory =
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GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-ogm-audio"));
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static GstPadTemplate *video_src_templ, *audio_src_templ;
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static GType gst_ogm_audio_parse_get_type (void);
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static GType gst_ogm_video_parse_get_type (void);
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static GType gst_ogm_parse_get_type (void);
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static void gst_ogm_audio_parse_base_init (GstOgmParseClass * klass);
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static void gst_ogm_video_parse_base_init (GstOgmParseClass * klass);
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static void gst_ogm_parse_class_init (GstOgmParseClass * klass);
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static void gst_ogm_parse_init (GstOgmParse * ogm);
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static const GstFormat *gst_ogm_parse_get_sink_formats (GstPad * pad);
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static gboolean gst_ogm_parse_sink_convert (GstPad * pad, GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
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static void gst_ogm_parse_chain (GstPad * pad, GstData * data);
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static GstElementStateReturn gst_ogm_parse_change_state (GstElement * element);
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GstElementClass *parent_class = NULL;
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static GType
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gst_ogm_parse_get_type (void)
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{
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static GType ogm_parse_type = 0;
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if (!ogm_parse_type) {
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static const GTypeInfo ogm_parse_info = {
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sizeof (GstOgmParseClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_ogm_parse_class_init,
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NULL,
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NULL,
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sizeof (GstOgmParse),
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0,
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(GInstanceInitFunc) gst_ogm_parse_init,
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};
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ogm_parse_type =
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g_type_register_static (GST_TYPE_ELEMENT,
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"GstOgmParse", &ogm_parse_info, 0);
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}
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return ogm_parse_type;
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}
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static GType
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gst_ogm_audio_parse_get_type (void)
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{
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static GType ogm_audio_parse_type = 0;
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if (!ogm_audio_parse_type) {
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static const GTypeInfo ogm_audio_parse_info = {
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sizeof (GstOgmParseClass),
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(GBaseInitFunc) gst_ogm_audio_parse_base_init,
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NULL,
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NULL,
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NULL,
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NULL,
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sizeof (GstOgmParse),
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0,
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NULL,
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};
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ogm_audio_parse_type =
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g_type_register_static (GST_TYPE_OGM_PARSE,
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"GstOgmAudioParse", &ogm_audio_parse_info, 0);
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}
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return ogm_audio_parse_type;
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}
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GType
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gst_ogm_video_parse_get_type (void)
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{
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static GType ogm_video_parse_type = 0;
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if (!ogm_video_parse_type) {
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static const GTypeInfo ogm_video_parse_info = {
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sizeof (GstOgmParseClass),
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(GBaseInitFunc) gst_ogm_video_parse_base_init,
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NULL,
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NULL,
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NULL,
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NULL,
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sizeof (GstOgmParse),
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0,
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NULL,
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};
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ogm_video_parse_type =
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g_type_register_static (GST_TYPE_OGM_PARSE,
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"GstOgmVideoParse", &ogm_video_parse_info, 0);
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}
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return ogm_video_parse_type;
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}
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static void
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gst_ogm_audio_parse_base_init (GstOgmParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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static GstElementDetails gst_ogm_audio_parse_details =
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GST_ELEMENT_DETAILS ("OGM audio stream parser",
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"Codec/Decoder/Audio",
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"parse an OGM audio header and stream",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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GstCaps *caps = gst_riff_create_audio_template_caps ();
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gst_element_class_set_details (element_class, &gst_ogm_audio_parse_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&ogm_audio_parse_sink_template_factory));
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audio_src_templ = gst_pad_template_new ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, caps);
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gst_element_class_add_pad_template (element_class, audio_src_templ);
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}
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static void
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gst_ogm_video_parse_base_init (GstOgmParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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static GstElementDetails gst_ogm_video_parse_details =
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GST_ELEMENT_DETAILS ("OGM video stream parser",
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"Codec/Decoder/Video",
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"parse an OGM video header and stream",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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GstCaps *caps = gst_riff_create_video_template_caps ();
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gst_element_class_set_details (element_class, &gst_ogm_video_parse_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&ogm_video_parse_sink_template_factory));
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video_src_templ = gst_pad_template_new ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, caps);
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gst_element_class_add_pad_template (element_class, video_src_templ);
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}
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static void
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gst_ogm_parse_class_init (GstOgmParseClass * klass)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gstelement_class->change_state = gst_ogm_parse_change_state;
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}
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static void
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gst_ogm_parse_init (GstOgmParse * ogm)
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{
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GstPadTemplate *templ;
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/* create the pads */
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templ = gst_static_pad_template_get (
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(G_OBJECT_TYPE (ogm) == GST_TYPE_OGM_AUDIO_PARSE) ?
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&ogm_audio_parse_sink_template_factory :
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&ogm_video_parse_sink_template_factory);
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ogm->sinkpad = gst_pad_new_from_template (templ, "sink");
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gst_pad_set_convert_function (ogm->sinkpad, gst_ogm_parse_sink_convert);
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gst_pad_set_formats_function (ogm->sinkpad, gst_ogm_parse_get_sink_formats);
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gst_pad_set_chain_function (ogm->sinkpad, gst_ogm_parse_chain);
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gst_element_add_pad (GST_ELEMENT (ogm), ogm->sinkpad);
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templ = (G_OBJECT_TYPE (ogm) == GST_TYPE_OGM_AUDIO_PARSE) ?
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audio_src_templ : video_src_templ;
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ogm->srcpad = gst_pad_new_from_template (templ, "src");
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gst_pad_use_explicit_caps (ogm->srcpad);
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gst_element_add_pad (GST_ELEMENT (ogm), ogm->srcpad);
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/* initalize */
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memset (&ogm->hdr, 0, sizeof (ogm->hdr));
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}
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static const GstFormat *
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gst_ogm_parse_get_sink_formats (GstPad * pad)
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{
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static GstFormat formats[] = {
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GST_FORMAT_DEFAULT,
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GST_FORMAT_TIME,
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0
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};
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return formats;
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}
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static gboolean
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gst_ogm_parse_sink_convert (GstPad * pad,
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GstFormat src_format, gint64 src_value,
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GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean res = FALSE;
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GstOgmParse *ogm = GST_OGM_PARSE (gst_pad_get_parent (pad));
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switch (src_format) {
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case GST_FORMAT_DEFAULT:
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switch (*dest_format) {
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case GST_FORMAT_TIME:
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switch (ogm->hdr.streamtype[0]) {
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case 'a':
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//*dest_value = ..;
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//res = TRUE;
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break;
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case 'v':
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*dest_value = (GST_SECOND / 10000000) *
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ogm->hdr.time_unit * src_value;
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res = TRUE;
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break;
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default:
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break;
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}
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break;
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default:
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break;
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}
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break;
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default:
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break;
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}
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return res;
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}
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static void
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gst_ogm_parse_chain (GstPad * pad, GstData * dat)
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{
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GstOgmParse *ogm = GST_OGM_PARSE (gst_pad_get_parent (pad));
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GstBuffer *buf = GST_BUFFER (dat);
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guint8 *data = GST_BUFFER_DATA (buf);
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guint size = GST_BUFFER_SIZE (buf);
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GST_DEBUG_OBJECT (ogm, "New packet with packet start code 0x%02x", data[0]);
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switch (data[0]) {
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case 0x01:{
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GstCaps *caps = NULL;
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/* stream header */
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if (size < sizeof (stream_header) + 1) {
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GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
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("Buffer too small"), (NULL));
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break;
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}
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if (!memcmp (&data[1], "video\000\000\000", 8)) {
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ogm->hdr.s.video.width = GST_READ_UINT32_LE (&data[45]);
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ogm->hdr.s.video.height = GST_READ_UINT32_LE (&data[49]);
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} else if (!memcmp (&data[1], "audio\000\000\000", 8)) {
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ogm->hdr.s.audio.channels = GST_READ_UINT32_LE (&data[45]);
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ogm->hdr.s.audio.blockalign = GST_READ_UINT32_LE (&data[47]);
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ogm->hdr.s.audio.avgbytespersec = GST_READ_UINT32_LE (&data[49]);
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} else {
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GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
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("Unknown stream type"), (NULL));
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break;
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}
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memcpy (ogm->hdr.streamtype, &data[1], 8);
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memcpy (ogm->hdr.subtype, &data[9], 4);
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ogm->hdr.size = GST_READ_UINT32_LE (&data[13]);
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ogm->hdr.time_unit = GST_READ_UINT64_LE (&data[17]);
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ogm->hdr.samples_per_unit = GST_READ_UINT64_LE (&data[25]);
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ogm->hdr.default_len = GST_READ_UINT32_LE (&data[33]);
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ogm->hdr.buffersize = GST_READ_UINT32_LE (&data[37]);
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ogm->hdr.bits_per_sample = GST_READ_UINT32_LE (&data[41]);
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switch (ogm->hdr.streamtype[0]) {
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case 'a':{
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caps = NULL;
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break;
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}
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case 'v':{
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guint32 fcc;
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fcc = GST_MAKE_FOURCC (ogm->hdr.subtype[0],
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ogm->hdr.subtype[1], ogm->hdr.subtype[2], ogm->hdr.subtype[3]);
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GST_LOG_OBJECT (ogm, "Type: %s, subtype: %" GST_FOURCC_FORMAT
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", size: %dx%d, timeunit: %" G_GINT64_FORMAT
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" (fps: %lf), s/u: %" G_GINT64_FORMAT ", "
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"def.len: %d, bufsize: %d, bps: %d",
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ogm->hdr.streamtype, GST_FOURCC_ARGS (fcc),
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ogm->hdr.s.video.width, ogm->hdr.s.video.height,
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ogm->hdr.time_unit, 10000000. / ogm->hdr.time_unit,
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ogm->hdr.samples_per_unit, ogm->hdr.default_len,
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ogm->hdr.buffersize, ogm->hdr.bits_per_sample);
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caps = gst_riff_create_video_caps (fcc, NULL, NULL, NULL);
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gst_caps_set_simple (caps,
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"width", G_TYPE_INT, ogm->hdr.s.video.width,
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"height", G_TYPE_INT, ogm->hdr.s.video.height,
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"framerate", G_TYPE_DOUBLE, 10000000. / ogm->hdr.time_unit, NULL);
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break;
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}
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default:
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g_assert_not_reached ();
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}
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if (!gst_pad_set_explicit_caps (ogm->srcpad, caps)) {
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GST_ELEMENT_ERROR (ogm, CORE, NEGOTIATION, (NULL), (NULL));
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}
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break;
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}
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case 0x03:
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/* comment - unused */
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break;
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default:
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if ((data[0] & 0x01) == 0) {
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/* data - push on */
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guint len = ((data[0] & 0xc0) >> 6) | ((data[0] & 0x02) << 1);
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guint xsize = 0;
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GstBuffer *sbuf;
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gboolean keyframe = (data[0] & 0x08) >> 3;
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if (size < len + 1) {
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GST_ELEMENT_ERROR (ogm, STREAM, WRONG_TYPE,
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("Buffer too small"), (NULL));
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break;
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}
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for (; len > 0; len--) {
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xsize = (xsize << 8) | data[len];
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}
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GST_DEBUG_OBJECT (ogm,
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"[0x%02x] Size of frame: %d, size of buffer: %d\n",
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data[0], xsize, size);
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/* ? */
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sbuf = gst_buffer_create_sub (buf, 1, size - 1);
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switch (ogm->hdr.streamtype[0]) {
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case 'v':
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if (keyframe)
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GST_BUFFER_FLAG_SET (sbuf, GST_BUFFER_KEY_UNIT);
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GST_BUFFER_TIMESTAMP (sbuf) = GST_BUFFER_TIMESTAMP (buf);
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GST_BUFFER_DURATION (sbuf) = (GST_SECOND / 10000000) *
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ogm->hdr.time_unit;
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break;
|
|
case 'a':
|
|
/* ? */
|
|
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:
|
|
memset (&ogm->hdr, 0, sizeof (ogm->hdr));
|
|
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_library_load ("riff") &&
|
|
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);
|
|
}
|