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670db4aade
Original commit message from CVS: * docs/plugins/Makefile.am: Add missing eol \ * docs/plugins/gst-plugins-bad-plugins-sections.txt: Place object names to standard sectionas plugin dont document those. * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: regenerate. * ext/ivorbis/vorbisdec.c: * ext/ivorbis/vorbisdec.h: Mark private vars and add short desc. * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: Add short desc.
1214 lines
33 KiB
C
1214 lines
33 KiB
C
/* GStreamer
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* Copyright (C) 2004 Benjamin Otte <in7y118@public.uni-hamburg.de>
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*
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* Tremor modifications <2006>:
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* Chris Lord, OpenedHand Ltd. <chris@openedhand.com>, http://www.o-hand.com/
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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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/**
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* SECTION:element-ivorbisdec
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* @short_description: a decoder that decodes Vorbis to raw audio
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* @see_also: vorbisenc, oggdemux
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*
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* <refsect2>
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* <para>
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* This element decodes a Vorbis stream to raw int audio.
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* <ulink url="http://www.vorbis.com/">Vorbis</ulink> is a royalty-free
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* audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
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* Foundation</ulink>.
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
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* </programlisting>
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* Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-03-01 (0.10.4)
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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 "vorbisdec.h"
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#include <string.h>
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#include <gst/audio/audio.h>
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#include <gst/tag/tag.h>
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#include <gst/audio/multichannel.h>
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GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug);
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#define GST_CAT_DEFAULT vorbisdec_debug
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static const GstElementDetails vorbis_dec_details =
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GST_ELEMENT_DETAILS ("Vorbis audio decoder",
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"Codec/Decoder/Audio",
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"decode raw vorbis streams to integer audio",
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"Benjamin Otte <in7y118@public.uni-hamburg.de>\n"
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"Chris Lord <chris@openedhand.com>");
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static GstStaticPadTemplate vorbis_dec_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 6 ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, " "depth = (int) 16, " "signed = (boolean) true")
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);
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static GstStaticPadTemplate vorbis_dec_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-vorbis")
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);
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GST_BOILERPLATE (GstIVorbisDec, gst_ivorbis_dec, GstElement, GST_TYPE_ELEMENT);
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static void vorbis_dec_finalize (GObject * object);
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static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer);
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static GstStateChangeReturn vorbis_dec_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event);
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static gboolean vorbis_dec_src_query (GstPad * pad, GstQuery * query);
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static gboolean vorbis_dec_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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static gboolean vorbis_dec_sink_query (GstPad * pad, GstQuery * query);
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static void
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gst_ivorbis_dec_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GstPadTemplate *src_template, *sink_template;
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src_template = gst_static_pad_template_get (&vorbis_dec_src_factory);
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gst_element_class_add_pad_template (element_class, src_template);
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sink_template = gst_static_pad_template_get (&vorbis_dec_sink_factory);
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gst_element_class_add_pad_template (element_class, sink_template);
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gst_element_class_set_details (element_class, &vorbis_dec_details);
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}
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static void
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gst_ivorbis_dec_class_init (GstIVorbisDecClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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gobject_class->finalize = vorbis_dec_finalize;
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (vorbis_dec_change_state);
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}
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static const GstQueryType *
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vorbis_get_query_types (GstPad * pad)
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{
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static const GstQueryType vorbis_dec_src_query_types[] = {
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GST_QUERY_POSITION,
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GST_QUERY_DURATION,
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GST_QUERY_CONVERT,
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0
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};
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return vorbis_dec_src_query_types;
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}
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static void
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gst_ivorbis_dec_init (GstIVorbisDec * dec, GstIVorbisDecClass * g_class)
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{
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dec->sinkpad = gst_pad_new_from_static_template (&vorbis_dec_sink_factory,
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"sink");
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gst_pad_set_event_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (vorbis_dec_sink_event));
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gst_pad_set_chain_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (vorbis_dec_chain));
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gst_pad_set_query_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (vorbis_dec_sink_query));
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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dec->srcpad = gst_pad_new_from_static_template (&vorbis_dec_src_factory,
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"src");
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gst_pad_set_event_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (vorbis_dec_src_event));
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gst_pad_set_query_type_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (vorbis_get_query_types));
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gst_pad_set_query_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (vorbis_dec_src_query));
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gst_pad_use_fixed_caps (dec->srcpad);
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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dec->queued = NULL;
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dec->pendingevents = NULL;
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dec->taglist = NULL;
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}
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static void
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vorbis_dec_finalize (GObject * object)
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{
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/* Release any possibly allocated libvorbis data.
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* _clear functions can safely be called multiple times
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*/
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GstIVorbisDec *vd = GST_IVORBIS_DEC (object);
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vorbis_block_clear (&vd->vb);
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vorbis_dsp_clear (&vd->vd);
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vorbis_comment_clear (&vd->vc);
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vorbis_info_clear (&vd->vi);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_ivorbis_dec_reset (GstIVorbisDec * dec)
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{
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GList *walk;
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dec->cur_timestamp = GST_CLOCK_TIME_NONE;
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dec->prev_timestamp = GST_CLOCK_TIME_NONE;
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dec->granulepos = -1;
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dec->discont = TRUE;
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gst_segment_init (&dec->segment, GST_FORMAT_TIME);
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for (walk = dec->queued; walk; walk = g_list_next (walk)) {
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gst_buffer_unref (GST_BUFFER_CAST (walk->data));
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}
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g_list_free (dec->queued);
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dec->queued = NULL;
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for (walk = dec->pendingevents; walk; walk = g_list_next (walk)) {
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gst_event_unref (GST_EVENT_CAST (walk->data));
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}
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g_list_free (dec->pendingevents);
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dec->pendingevents = NULL;
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if (dec->taglist)
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gst_tag_list_free (dec->taglist);
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dec->taglist = NULL;
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}
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static gboolean
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vorbis_dec_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 = TRUE;
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GstIVorbisDec *dec;
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guint64 scale = 1;
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if (src_format == *dest_format) {
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*dest_value = src_value;
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return TRUE;
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}
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dec = GST_IVORBIS_DEC (gst_pad_get_parent (pad));
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if (!dec->initialized)
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goto no_header;
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if (dec->sinkpad == pad &&
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(src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES))
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goto no_format;
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switch (src_format) {
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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scale = sizeof (gint32) * dec->vi.channels;
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case GST_FORMAT_DEFAULT:
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*dest_value =
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scale * gst_util_uint64_scale_int (src_value, dec->vi.rate,
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GST_SECOND);
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break;
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_DEFAULT:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = src_value * sizeof (gint32) * dec->vi.channels;
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break;
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case GST_FORMAT_TIME:
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*dest_value =
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gst_util_uint64_scale_int (src_value, GST_SECOND, dec->vi.rate);
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break;
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_DEFAULT:
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*dest_value = src_value / (sizeof (gint32) * dec->vi.channels);
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break;
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case GST_FORMAT_TIME:
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*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
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dec->vi.rate * sizeof (gint32) * dec->vi.channels);
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break;
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default:
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res = FALSE;
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}
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break;
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default:
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res = FALSE;
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}
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done:
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gst_object_unref (dec);
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return res;
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/* ERRORS */
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no_header:
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{
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GST_DEBUG_OBJECT (dec, "no header packets received");
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res = FALSE;
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goto done;
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}
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no_format:
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{
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GST_DEBUG_OBJECT (dec, "formats unsupported");
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res = FALSE;
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goto done;
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}
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}
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static gboolean
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vorbis_dec_src_query (GstPad * pad, GstQuery * query)
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{
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GstIVorbisDec *dec;
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gboolean res = FALSE;
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dec = GST_IVORBIS_DEC (gst_pad_get_parent (pad));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_POSITION:
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{
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gint64 granulepos, value;
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GstFormat my_format, format;
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gint64 time;
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/* we start from the last seen granulepos */
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granulepos = dec->granulepos;
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gst_query_parse_position (query, &format, NULL);
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/* and convert to the final format in two steps with time as the
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* intermediate step */
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my_format = GST_FORMAT_TIME;
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if (!(res =
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vorbis_dec_convert (pad, GST_FORMAT_DEFAULT, granulepos,
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&my_format, &time)))
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goto error;
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/* correct for the segment values */
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time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
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GST_LOG_OBJECT (dec,
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"query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));
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/* and convert to the final format */
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if (!(res = vorbis_dec_convert (pad, my_format, time, &format, &value)))
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goto error;
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gst_query_set_position (query, format, value);
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GST_LOG_OBJECT (dec,
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"query %p: we return %lld (format %u)", query, value, format);
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break;
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}
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case GST_QUERY_DURATION:
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{
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GstPad *peer;
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|
|
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if (!(peer = gst_pad_get_peer (dec->sinkpad))) {
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GST_WARNING_OBJECT (dec, "sink pad %" GST_PTR_FORMAT " is not linked",
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dec->sinkpad);
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goto error;
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}
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|
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res = gst_pad_query (peer, query);
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gst_object_unref (peer);
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if (!res)
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goto error;
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break;
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}
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case GST_QUERY_CONVERT:
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{
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GstFormat src_fmt, dest_fmt;
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gint64 src_val, dest_val;
|
|
|
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
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if (!(res =
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vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val)))
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goto error;
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
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|
break;
|
|
}
|
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done:
|
|
gst_object_unref (dec);
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|
|
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return res;
|
|
|
|
/* ERRORS */
|
|
error:
|
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{
|
|
GST_WARNING_OBJECT (dec, "error handling query");
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goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
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vorbis_dec_sink_query (GstPad * pad, GstQuery * query)
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|
{
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GstIVorbisDec *dec;
|
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gboolean res;
|
|
|
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dec = GST_IVORBIS_DEC (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
|
|
if (!(res =
|
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vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val)))
|
|
goto error;
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
|
|
done:
|
|
gst_object_unref (dec);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
GST_DEBUG_OBJECT (dec, "error converting value");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
vorbis_dec_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstIVorbisDec *dec;
|
|
|
|
dec = GST_IVORBIS_DEC (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
GstFormat format, tformat;
|
|
gdouble rate;
|
|
GstEvent *real_seek;
|
|
GstSeekFlags flags;
|
|
GstSeekType cur_type, stop_type;
|
|
gint64 cur, stop;
|
|
gint64 tcur, tstop;
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
|
|
&stop_type, &stop);
|
|
gst_event_unref (event);
|
|
|
|
/* we have to ask our peer to seek to time here as we know
|
|
* nothing about how to generate a granulepos from the src
|
|
* formats or anything.
|
|
*
|
|
* First bring the requested format to time
|
|
*/
|
|
tformat = GST_FORMAT_TIME;
|
|
if (!(res = vorbis_dec_convert (pad, format, cur, &tformat, &tcur)))
|
|
goto convert_error;
|
|
if (!(res = vorbis_dec_convert (pad, format, stop, &tformat, &tstop)))
|
|
goto convert_error;
|
|
|
|
/* then seek with time on the peer */
|
|
real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
|
|
flags, cur_type, tcur, stop_type, tstop);
|
|
|
|
res = gst_pad_push_event (dec->sinkpad, real_seek);
|
|
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_push_event (dec->sinkpad, event);
|
|
break;
|
|
}
|
|
done:
|
|
gst_object_unref (dec);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
convert_error:
|
|
{
|
|
GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
vorbis_dec_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
gboolean ret = FALSE;
|
|
GstIVorbisDec *dec;
|
|
|
|
dec = GST_IVORBIS_DEC (gst_pad_get_parent (pad));
|
|
|
|
GST_LOG_OBJECT (dec, "handling event");
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
ret = gst_pad_push_event (dec->srcpad, event);
|
|
break;
|
|
case GST_EVENT_FLUSH_START:
|
|
ret = gst_pad_push_event (dec->srcpad, event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* here we must clean any state in the decoder */
|
|
#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
|
|
vorbis_synthesis_restart (&dec->vd);
|
|
#endif
|
|
gst_ivorbis_dec_reset (dec);
|
|
ret = gst_pad_push_event (dec->srcpad, event);
|
|
break;
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
GstFormat format;
|
|
gdouble rate, arate;
|
|
gint64 start, stop, time;
|
|
gboolean update;
|
|
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
|
|
&start, &stop, &time);
|
|
|
|
/* we need time and a positive rate for now */
|
|
if (format != GST_FORMAT_TIME)
|
|
goto newseg_wrong_format;
|
|
|
|
if (rate <= 0.0)
|
|
goto newseg_wrong_rate;
|
|
|
|
GST_DEBUG_OBJECT (dec,
|
|
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
|
|
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
|
|
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
|
|
GST_TIME_ARGS (time));
|
|
|
|
/* now configure the values */
|
|
gst_segment_set_newsegment_full (&dec->segment, update,
|
|
rate, arate, format, start, stop, time);
|
|
|
|
if (dec->initialized)
|
|
/* and forward */
|
|
ret = gst_pad_push_event (dec->srcpad, event);
|
|
else {
|
|
/* store it to send once we're initialized */
|
|
dec->pendingevents = g_list_append (dec->pendingevents, event);
|
|
ret = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_push_event (dec->srcpad, event);
|
|
break;
|
|
}
|
|
done:
|
|
gst_object_unref (dec);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
newseg_wrong_format:
|
|
{
|
|
GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
|
|
goto done;
|
|
}
|
|
newseg_wrong_rate:
|
|
{
|
|
GST_DEBUG_OBJECT (dec, "negative rates not supported yet");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_handle_identification_packet (GstIVorbisDec * vd)
|
|
{
|
|
GstCaps *caps;
|
|
const GstAudioChannelPosition *pos = NULL;
|
|
|
|
switch (vd->vi.channels) {
|
|
case 1:
|
|
case 2:
|
|
/* nothing */
|
|
break;
|
|
case 3:{
|
|
static const GstAudioChannelPosition pos3[] = {
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
|
|
};
|
|
pos = pos3;
|
|
break;
|
|
}
|
|
case 4:{
|
|
static const GstAudioChannelPosition pos4[] = {
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
|
|
};
|
|
pos = pos4;
|
|
break;
|
|
}
|
|
case 5:{
|
|
static const GstAudioChannelPosition pos5[] = {
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
|
|
};
|
|
pos = pos5;
|
|
break;
|
|
}
|
|
case 6:{
|
|
static const GstAudioChannelPosition pos6[] = {
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
GST_AUDIO_CHANNEL_POSITION_LFE
|
|
};
|
|
pos = pos6;
|
|
break;
|
|
}
|
|
default:
|
|
goto channel_count_error;
|
|
}
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw-int",
|
|
"rate", G_TYPE_INT, vd->vi.rate,
|
|
"channels", G_TYPE_INT, vd->vi.channels,
|
|
"endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32,
|
|
"depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
|
|
if (pos) {
|
|
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
|
|
}
|
|
gst_pad_set_caps (vd->srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERROR */
|
|
channel_count_error:
|
|
{
|
|
GST_ELEMENT_ERROR (vd, STREAM, NOT_IMPLEMENTED, (NULL),
|
|
("Unsupported channel count %d", vd->vi.channels));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_handle_comment_packet (GstIVorbisDec * vd, ogg_packet * packet)
|
|
{
|
|
#if 0
|
|
guint bitrate = 0;
|
|
gchar *encoder = NULL;
|
|
GstTagList *list;
|
|
GstBuffer *buf;
|
|
|
|
GST_DEBUG_OBJECT (vd, "parsing comment packet");
|
|
|
|
buf = gst_buffer_new_and_alloc (packet->bytes);
|
|
GST_BUFFER_DATA (buf) = packet->packet;
|
|
|
|
list =
|
|
gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7,
|
|
&encoder);
|
|
|
|
vd->taglist = gst_tag_list_merge (vd->taglist, list, GST_TAG_MERGE_REPLACE);
|
|
|
|
gst_tag_list_free (list);
|
|
gst_buffer_unref (buf);
|
|
|
|
if (!vd->taglist) {
|
|
GST_ERROR_OBJECT (vd, "couldn't decode comments");
|
|
vd->taglist = gst_tag_list_new ();
|
|
}
|
|
if (encoder) {
|
|
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_ENCODER, encoder, NULL);
|
|
g_free (encoder);
|
|
}
|
|
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_ENCODER_VERSION, vd->vi.version,
|
|
GST_TAG_AUDIO_CODEC, "Vorbis", NULL);
|
|
if (vd->vi.bitrate_nominal > 0) {
|
|
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_NOMINAL_BITRATE, (guint) vd->vi.bitrate_nominal, NULL);
|
|
bitrate = vd->vi.bitrate_nominal;
|
|
}
|
|
if (vd->vi.bitrate_upper > 0) {
|
|
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, NULL);
|
|
if (!bitrate)
|
|
bitrate = vd->vi.bitrate_upper;
|
|
}
|
|
if (vd->vi.bitrate_lower > 0) {
|
|
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL);
|
|
if (!bitrate)
|
|
bitrate = vd->vi.bitrate_lower;
|
|
}
|
|
if (bitrate) {
|
|
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_BITRATE, (guint) bitrate, NULL);
|
|
}
|
|
|
|
if (vd->initialized) {
|
|
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad,
|
|
vd->taglist);
|
|
vd->taglist = NULL;
|
|
} else {
|
|
/* Only post them as messages for the time being. *
|
|
* They will be pushed on the pad once the decoder is initialized */
|
|
gst_element_post_message (GST_ELEMENT_CAST (vd),
|
|
gst_message_new_tag (GST_OBJECT (vd), gst_tag_list_copy (vd->taglist)));
|
|
}
|
|
#endif
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_handle_type_packet (GstIVorbisDec * vd)
|
|
{
|
|
GList *walk;
|
|
|
|
g_assert (vd->initialized == FALSE);
|
|
|
|
vorbis_synthesis_init (&vd->vd, &vd->vi);
|
|
vorbis_block_init (&vd->vd, &vd->vb);
|
|
vd->initialized = TRUE;
|
|
|
|
if (vd->pendingevents) {
|
|
for (walk = vd->pendingevents; walk; walk = g_list_next (walk))
|
|
gst_pad_push_event (vd->srcpad, GST_EVENT_CAST (walk->data));
|
|
g_list_free (vd->pendingevents);
|
|
vd->pendingevents = NULL;
|
|
}
|
|
|
|
if (vd->taglist) {
|
|
/* The tags have already been sent on the bus as messages. */
|
|
gst_pad_push_event (vd->srcpad, gst_event_new_tag (vd->taglist));
|
|
vd->taglist = NULL;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_handle_header_packet (GstIVorbisDec * vd, ogg_packet * packet)
|
|
{
|
|
GstFlowReturn res;
|
|
|
|
GST_DEBUG_OBJECT (vd, "parsing header packet");
|
|
|
|
/* Packetno = 0 if the first byte is exactly 0x01 */
|
|
packet->b_o_s = packet->packet->length ?
|
|
((packet->packet->buffer->data[0] == 0x1) ? 1 : 0) : 0;
|
|
|
|
if (vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet))
|
|
goto header_read_error;
|
|
|
|
switch (packet->packet->length ? packet->packet->buffer->data[0] : 0x0) {
|
|
case 0x01:
|
|
res = vorbis_handle_identification_packet (vd);
|
|
break;
|
|
case 0x03:
|
|
res = vorbis_handle_comment_packet (vd, packet);
|
|
break;
|
|
case 0x05:
|
|
res = vorbis_handle_type_packet (vd);
|
|
break;
|
|
default:
|
|
/* ignore */
|
|
g_warning ("unknown vorbis header packet found");
|
|
res = GST_FLOW_OK;
|
|
break;
|
|
}
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
header_read_error:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("couldn't read header packet"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/* Taken from Tremor, misc.h */
|
|
#ifdef _ARM_ASSEM_
|
|
static inline ogg_int32_t
|
|
CLIP_TO_15 (ogg_int32_t x)
|
|
{
|
|
int tmp;
|
|
asm volatile ("subs %1, %0, #32768\n\t"
|
|
"movpl %0, #0x7f00\n\t"
|
|
"orrpl %0, %0, #0xff\n"
|
|
"adds %1, %0, #32768\n\t"
|
|
"movmi %0, #0x8000":"+r" (x), "=r" (tmp)
|
|
::"cc");
|
|
|
|
return (x);
|
|
}
|
|
#else
|
|
static inline ogg_int32_t
|
|
CLIP_TO_15 (ogg_int32_t x)
|
|
{
|
|
int ret = x;
|
|
|
|
ret -= ((x <= 32767) - 1) & (x - 32767);
|
|
ret -= ((x >= -32768) - 1) & (x + 32768);
|
|
return (ret);
|
|
}
|
|
#endif
|
|
|
|
static void
|
|
copy_samples (gint32 * out, ogg_int32_t ** in, guint samples, gint channels)
|
|
{
|
|
gint i, j;
|
|
|
|
for (j = 0; j < samples; j++) {
|
|
for (i = 0; i < channels; i++) {
|
|
*out++ = CLIP_TO_15 (in[i][j] >> 9);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* clip output samples to the segment boundaries
|
|
*/
|
|
static gboolean
|
|
vorbis_do_clip (GstIVorbisDec * dec, GstBuffer * buf)
|
|
{
|
|
gint64 start, stop, cstart, cstop, diff;
|
|
|
|
start = GST_BUFFER_TIMESTAMP (buf);
|
|
stop = start + GST_BUFFER_DURATION (buf);
|
|
|
|
if (!gst_segment_clip (&dec->segment, GST_FORMAT_TIME,
|
|
start, stop, &cstart, &cstop))
|
|
goto clipped;
|
|
|
|
/* see if some clipping happened */
|
|
diff = cstart - start;
|
|
if (diff > 0) {
|
|
GST_BUFFER_TIMESTAMP (buf) = cstart;
|
|
GST_BUFFER_DURATION (buf) -= diff;
|
|
|
|
/* bring clipped time to samples */
|
|
diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND);
|
|
/* samples to bytes */
|
|
diff *= (sizeof (gint32) * dec->vi.channels);
|
|
GST_DEBUG_OBJECT (dec, "clipping start to %" GST_TIME_FORMAT " %"
|
|
G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstart), diff);
|
|
GST_BUFFER_DATA (buf) += diff;
|
|
GST_BUFFER_SIZE (buf) -= diff;
|
|
}
|
|
diff = stop - cstop;
|
|
if (diff > 0) {
|
|
GST_BUFFER_DURATION (buf) -= diff;
|
|
|
|
/* bring clipped time to samples and then to bytes */
|
|
diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND);
|
|
diff *= (sizeof (gint32) * dec->vi.channels);
|
|
GST_DEBUG_OBJECT (dec, "clipping stop to %" GST_TIME_FORMAT " %"
|
|
G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff);
|
|
GST_BUFFER_SIZE (buf) -= diff;
|
|
}
|
|
|
|
return FALSE;
|
|
|
|
/* dropped buffer */
|
|
clipped:
|
|
{
|
|
GST_DEBUG_OBJECT (dec, "clipped buffer");
|
|
gst_buffer_unref (buf);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_dec_push (GstIVorbisDec * dec, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn result;
|
|
gint64 outoffset = GST_BUFFER_OFFSET (buf);
|
|
|
|
if (outoffset == -1) {
|
|
dec->queued = g_list_append (dec->queued, buf);
|
|
GST_DEBUG_OBJECT (dec, "queued buffer");
|
|
result = GST_FLOW_OK;
|
|
} else {
|
|
if (G_UNLIKELY (dec->queued)) {
|
|
gint64 size;
|
|
GList *walk;
|
|
|
|
GST_DEBUG_OBJECT (dec, "first buffer with offset %lld", outoffset);
|
|
|
|
size = g_list_length (dec->queued);
|
|
for (walk = g_list_last (dec->queued); walk;
|
|
walk = g_list_previous (walk)) {
|
|
GstBuffer *buffer = GST_BUFFER (walk->data);
|
|
|
|
outoffset -=
|
|
GST_BUFFER_SIZE (buffer) / (sizeof (gint32) * dec->vi.channels);
|
|
|
|
GST_BUFFER_OFFSET (buffer) = outoffset;
|
|
GST_BUFFER_TIMESTAMP (buffer) =
|
|
gst_util_uint64_scale_int (outoffset, GST_SECOND, dec->vi.rate);
|
|
GST_DEBUG_OBJECT (dec, "patch buffer %" G_GUINT64_FORMAT
|
|
" offset %" G_GUINT64_FORMAT, size, outoffset);
|
|
size--;
|
|
}
|
|
for (walk = dec->queued; walk; walk = g_list_next (walk)) {
|
|
GstBuffer *buffer = GST_BUFFER (walk->data);
|
|
|
|
/* clips or returns FALSE with buffer unreffed when completely
|
|
* clipped */
|
|
if (vorbis_do_clip (dec, buffer))
|
|
continue;
|
|
|
|
if (dec->discont) {
|
|
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
|
|
dec->discont = FALSE;
|
|
}
|
|
/* ignore the result */
|
|
gst_pad_push (dec->srcpad, buffer);
|
|
}
|
|
g_list_free (dec->queued);
|
|
dec->queued = NULL;
|
|
}
|
|
|
|
/* clip */
|
|
if (vorbis_do_clip (dec, buf))
|
|
return GST_FLOW_OK;
|
|
|
|
if (dec->discont) {
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
dec->discont = FALSE;
|
|
}
|
|
result = gst_pad_push (dec->srcpad, buf);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_handle_data_packet (GstIVorbisDec * vd, ogg_packet * packet)
|
|
{
|
|
ogg_int32_t **pcm;
|
|
guint sample_count;
|
|
GstBuffer *out;
|
|
GstFlowReturn result;
|
|
gint size;
|
|
|
|
if (!vd->initialized)
|
|
goto not_initialized;
|
|
|
|
/* FIXME, we should queue undecoded packets here until we get
|
|
* a timestamp, then we reverse timestamp the queued packets and
|
|
* clip them, then we decode only the ones we want and don't
|
|
* keep decoded data in memory.
|
|
* Ideally, of course, the demuxer gives us a valid timestamp on
|
|
* the first packet.
|
|
*/
|
|
|
|
/* normal data packet */
|
|
/* FIXME, we can skip decoding if the packet is outside of the
|
|
* segment, this is however not very trivial as we need a previous
|
|
* packet to decode the current one so we must be carefull not to
|
|
* throw away too much. For now we decode everything and clip right
|
|
* before pushing data. */
|
|
if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet, 1)))
|
|
goto could_not_read;
|
|
|
|
if (G_UNLIKELY (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0))
|
|
goto not_accepted;
|
|
|
|
/* assume all goes well here */
|
|
result = GST_FLOW_OK;
|
|
|
|
/* count samples ready for reading */
|
|
if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
|
|
goto done;
|
|
|
|
size = sample_count * vd->vi.channels * sizeof (gint32);
|
|
|
|
/* alloc buffer for it */
|
|
result =
|
|
gst_pad_alloc_buffer_and_set_caps (vd->srcpad, GST_BUFFER_OFFSET_NONE,
|
|
size, GST_PAD_CAPS (vd->srcpad), &out);
|
|
if (G_UNLIKELY (result != GST_FLOW_OK))
|
|
goto done;
|
|
|
|
/* get samples ready for reading now, should be sample_count */
|
|
if (G_UNLIKELY ((vorbis_synthesis_pcmout (&vd->vd, &pcm)) != sample_count))
|
|
goto wrong_samples;
|
|
|
|
/* copy samples in buffer */
|
|
copy_samples ((gint32 *) GST_BUFFER_DATA (out), pcm, sample_count,
|
|
vd->vi.channels);
|
|
|
|
GST_BUFFER_SIZE (out) = size;
|
|
GST_BUFFER_OFFSET (out) = vd->granulepos;
|
|
if (vd->granulepos != -1) {
|
|
GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count;
|
|
GST_BUFFER_TIMESTAMP (out) =
|
|
gst_util_uint64_scale_int (vd->granulepos, GST_SECOND, vd->vi.rate);
|
|
} else {
|
|
GST_BUFFER_TIMESTAMP (out) = -1;
|
|
}
|
|
/* this should not overflow */
|
|
GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate;
|
|
|
|
if (vd->cur_timestamp != GST_CLOCK_TIME_NONE) {
|
|
GST_BUFFER_TIMESTAMP (out) = vd->cur_timestamp;
|
|
GST_DEBUG_OBJECT (vd,
|
|
"cur_timestamp: %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT " = % "
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (vd->cur_timestamp),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (out)),
|
|
GST_TIME_ARGS (vd->cur_timestamp + GST_BUFFER_DURATION (out)));
|
|
vd->cur_timestamp += GST_BUFFER_DURATION (out);
|
|
GST_BUFFER_OFFSET (out) = GST_CLOCK_TIME_TO_FRAMES (vd->cur_timestamp,
|
|
vd->vi.rate);
|
|
GST_BUFFER_OFFSET_END (out) = GST_BUFFER_OFFSET (out) + sample_count;
|
|
}
|
|
|
|
if (vd->granulepos != -1)
|
|
vd->granulepos += sample_count;
|
|
|
|
result = vorbis_dec_push (vd, out);
|
|
|
|
done:
|
|
vorbis_synthesis_read (&vd->vd, sample_count);
|
|
|
|
/* granulepos is the last sample in the packet */
|
|
if (packet->granulepos != -1)
|
|
vd->granulepos = packet->granulepos;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
not_initialized:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("no header sent yet"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
could_not_read:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("couldn't read data packet"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
not_accepted:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("vorbis decoder did not accept data packet"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
wrong_samples:
|
|
{
|
|
gst_buffer_unref (out);
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("vorbis decoder reported wrong number of samples"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_dec_chain (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstIVorbisDec *vd;
|
|
ogg_packet packet;
|
|
ogg_reference ref;
|
|
ogg_buffer buf;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
GstClockTime timestamp;
|
|
guint64 offset_end;
|
|
|
|
vd = GST_IVORBIS_DEC (gst_pad_get_parent (pad));
|
|
|
|
/* resync on DISCONT */
|
|
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT))) {
|
|
GST_DEBUG_OBJECT (vd, "received DISCONT buffer");
|
|
vd->granulepos = -1;
|
|
vd->cur_timestamp = GST_CLOCK_TIME_NONE;
|
|
vd->prev_timestamp = GST_CLOCK_TIME_NONE;
|
|
#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
|
|
vorbis_synthesis_restart (&vd->vd);
|
|
#endif
|
|
vd->discont = TRUE;
|
|
}
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
offset_end = GST_BUFFER_OFFSET_END (buffer);
|
|
|
|
/* only ogg has granulepos, demuxers of other container formats
|
|
* might provide us with timestamps instead (e.g. matroskademux) */
|
|
if (offset_end == GST_BUFFER_OFFSET_NONE && timestamp != GST_CLOCK_TIME_NONE) {
|
|
/* we might get multiple consecutive buffers with the same timestamp */
|
|
if (timestamp != vd->prev_timestamp) {
|
|
vd->cur_timestamp = timestamp;
|
|
vd->prev_timestamp = timestamp;
|
|
}
|
|
} else {
|
|
vd->cur_timestamp = GST_CLOCK_TIME_NONE;
|
|
vd->prev_timestamp = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
/* make ogg_packet out of the buffer */
|
|
buf.data = GST_BUFFER_DATA (buffer);
|
|
buf.size = GST_BUFFER_SIZE (buffer);
|
|
buf.refcount = 1;
|
|
buf.ptr.owner = NULL;
|
|
buf.ptr.next = NULL;
|
|
|
|
ref.buffer = &buf;
|
|
ref.begin = 0;
|
|
ref.length = buf.size;
|
|
ref.next = NULL;
|
|
|
|
packet.packet = &ref;
|
|
packet.bytes = ref.length;
|
|
packet.granulepos = offset_end;
|
|
packet.packetno = 0; /* we don't care */
|
|
/*
|
|
* FIXME. Is there anyway to know that this is the last packet and
|
|
* set e_o_s??
|
|
* Yes there is, keep one packet at all times and only push out when
|
|
* you receive a new one. Implement this.
|
|
*/
|
|
packet.e_o_s = 0;
|
|
|
|
if (G_UNLIKELY (packet.bytes < 1))
|
|
goto wrong_size;
|
|
|
|
GST_DEBUG_OBJECT (vd, "vorbis granule: %" G_GINT64_FORMAT,
|
|
(gint64) packet.granulepos);
|
|
|
|
/* switch depending on packet type */
|
|
if (buf.data[0] & 1) {
|
|
if (vd->initialized) {
|
|
GST_WARNING_OBJECT (vd, "Already initialized, so ignoring header packet");
|
|
goto done;
|
|
}
|
|
result = vorbis_handle_header_packet (vd, &packet);
|
|
} else {
|
|
result = vorbis_handle_data_packet (vd, &packet);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (vd, "offset end: %" G_GUINT64_FORMAT, offset_end);
|
|
|
|
done:
|
|
gst_buffer_unref (buffer);
|
|
gst_object_unref (vd);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_size:
|
|
{
|
|
GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty buffer received"));
|
|
result = GST_FLOW_ERROR;
|
|
vd->discont = TRUE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
vorbis_dec_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstIVorbisDec *vd = GST_IVORBIS_DEC (element);
|
|
GstStateChangeReturn res;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
vorbis_info_init (&vd->vi);
|
|
vorbis_comment_init (&vd->vc);
|
|
vd->initialized = FALSE;
|
|
gst_ivorbis_dec_reset (vd);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
res = parent_class->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
GST_DEBUG_OBJECT (vd, "PAUSED -> READY, clearing vorbis structures");
|
|
vorbis_block_clear (&vd->vb);
|
|
vorbis_dsp_clear (&vd->vd);
|
|
vorbis_comment_clear (&vd->vc);
|
|
vorbis_info_clear (&vd->vi);
|
|
gst_ivorbis_dec_reset (vd);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|