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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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3a0a2898af
Original commit message from CVS: * ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push), (gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init): * ext/alsa/gstalsa.c: (gst_alsa_get_caps): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/dts/gstdtsdec.c: (gst_dtsdec_channels), (gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init): * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst), (gst_faad_chanpos_to_gst), (gst_faad_sinkconnect), (gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain), (gst_faad_change_state), (plugin_init): * ext/faad/gstfaad.h: * ext/vorbis/vorbis.c: (plugin_init): * ext/vorbis/vorbisdec.c: (vorbis_dec_chain): * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/audio.c: (plugin_init): * gst-libs/gst/audio/multichannel.c: (gst_audio_check_channel_positions), (gst_audio_get_channel_positions), (gst_audio_set_channel_positions), (gst_audio_set_structure_channel_positions_list), (add_list_to_struct), (gst_audio_set_caps_channel_positions_list), (gst_audio_fixate_channel_positions): * gst-libs/gst/audio/multichannel.h: * gst-libs/gst/audio/testchannels.c: (main): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_dispose), (gst_audio_convert_getcaps), (gst_audio_convert_parse_caps), (gst_audio_convert_link), (gst_audio_convert_fixate), (gst_audio_convert_channels): * gst/audioconvert/plugin.c: (plugin_init): Surround sound support.
594 lines
18 KiB
C
594 lines
18 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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* 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 "vorbisdec.h"
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#include <string.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 GstElementDetails vorbis_dec_details = {
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"VorbisDec",
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"Codec/Decoder/Audio",
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"decode raw vorbis streams to float audio",
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"Benjamin Otte <in7y118@public.uni-hamburg.de>",
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};
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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};
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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-float, "
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"rate = (int) [ 8000, 50000 ], "
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"channels = (int) [ 1, 6 ], " "endianness = (int) BYTE_ORDER, "
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/* no ifdef in macros, please
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#ifdef GST_VORBIS_DEC_SEQUENTIAL
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"layout = \"sequential\", "
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#endif
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*/
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"width = (int) 32, " "buffer-frames = (int) 0")
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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 (GstVorbisDec, gst_vorbis_dec, GstElement, GST_TYPE_ELEMENT);
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static void vorbis_dec_chain (GstPad * pad, GstData * data);
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static GstElementStateReturn vorbis_dec_change_state (GstElement * element);
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static const GstFormat *vorbis_dec_get_formats (GstPad * pad);
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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,
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GstQueryType query, GstFormat * format, gint64 * value);
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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 void
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gst_vorbis_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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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&vorbis_dec_src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&vorbis_dec_sink_factory));
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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_vorbis_dec_class_init (GstVorbisDecClass * klass)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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gstelement_class->change_state = vorbis_dec_change_state;
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}
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static const GstFormat *
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vorbis_dec_get_formats (GstPad * pad)
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{
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static GstFormat src_formats[] = {
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GST_FORMAT_BYTES,
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GST_FORMAT_DEFAULT, /* samples in the audio case */
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GST_FORMAT_TIME,
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0
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};
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static GstFormat sink_formats[] = {
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/*GST_FORMAT_BYTES, */
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GST_FORMAT_TIME,
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GST_FORMAT_DEFAULT, /* granulepos or samples */
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0
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};
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return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats);
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}
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static const GstEventMask *
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vorbis_get_event_masks (GstPad * pad)
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{
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static const GstEventMask vorbis_dec_src_event_masks[] = {
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{GST_EVENT_SEEK, GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH},
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{0,}
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};
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return vorbis_dec_src_event_masks;
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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_TOTAL,
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GST_QUERY_POSITION,
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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_vorbis_dec_init (GstVorbisDec * dec)
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{
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dec->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&vorbis_dec_sink_factory), "sink");
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gst_pad_set_chain_function (dec->sinkpad, vorbis_dec_chain);
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gst_pad_set_formats_function (dec->sinkpad, vorbis_dec_get_formats);
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gst_pad_set_convert_function (dec->sinkpad, vorbis_dec_convert);
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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dec->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&vorbis_dec_src_factory), "src");
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gst_pad_use_explicit_caps (dec->srcpad);
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gst_pad_set_event_mask_function (dec->srcpad, vorbis_get_event_masks);
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gst_pad_set_event_function (dec->srcpad, vorbis_dec_src_event);
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gst_pad_set_query_type_function (dec->srcpad, vorbis_get_query_types);
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gst_pad_set_query_function (dec->srcpad, vorbis_dec_src_query);
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gst_pad_set_formats_function (dec->srcpad, vorbis_dec_get_formats);
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gst_pad_set_convert_function (dec->srcpad, vorbis_dec_convert);
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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GST_FLAG_SET (dec, GST_ELEMENT_EVENT_AWARE);
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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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GstVorbisDec *dec;
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guint64 scale = 1;
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dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
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if (dec->packetno < 1)
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return FALSE;
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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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return FALSE;
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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 (float) * dec->vi.channels;
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case GST_FORMAT_DEFAULT:
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*dest_value = scale * (src_value * dec->vi.rate / 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 (float) * dec->vi.channels;
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break;
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case GST_FORMAT_TIME:
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*dest_value = 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 (float) * dec->vi.channels);
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break;
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case GST_FORMAT_TIME:
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*dest_value = src_value * GST_SECOND /
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(dec->vi.rate * sizeof (float) * 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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return res;
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}
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static gboolean
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vorbis_dec_src_query (GstPad * pad, GstQueryType query, GstFormat * format,
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gint64 * value)
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{
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gint64 granulepos = 0;
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GstVorbisDec *dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
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GstFormat my_format = GST_FORMAT_DEFAULT;
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if (query == GST_QUERY_POSITION) {
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granulepos = dec->granulepos;
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} else {
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/* query peer in default format */
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if (!dec->sinkpad ||
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!gst_pad_query (GST_PAD_PEER (dec->sinkpad), query, &my_format,
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&granulepos))
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return FALSE;
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}
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/* and convert to the final format */
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if (!gst_pad_convert (pad, GST_FORMAT_DEFAULT, granulepos, format, value))
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return FALSE;
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GST_LOG_OBJECT (dec,
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"query %u: peer returned granulepos: %llu - we return %llu (format %u)\n",
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query, granulepos, *value, *format);
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return TRUE;
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}
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static gboolean
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vorbis_dec_src_event (GstPad * pad, GstEvent * event)
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{
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gboolean res = TRUE;
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GstVorbisDec *dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_SEEK:{
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guint64 value;
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GstFormat my_format = GST_FORMAT_DEFAULT;
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/* convert to granulepos */
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res = vorbis_dec_convert (pad, GST_EVENT_SEEK_FORMAT (event),
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GST_EVENT_SEEK_OFFSET (event), &my_format, &value);
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if (res) {
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GstEvent *real_seek = gst_event_new_seek (
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(GST_EVENT_SEEK_TYPE (event) & ~GST_SEEK_FORMAT_MASK) |
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GST_FORMAT_DEFAULT,
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value);
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res = gst_pad_send_event (GST_PAD_PEER (dec->sinkpad), real_seek);
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}
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gst_event_unref (event);
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break;
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}
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default:
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res = gst_pad_event_default (pad, event);
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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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vorbis_dec_event (GstVorbisDec * dec, GstEvent * event)
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{
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guint64 value, time, bytes;
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GST_LOG_OBJECT (dec, "handling event");
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_DISCONTINUOUS:
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if (gst_event_discont_get_value (event, GST_FORMAT_DEFAULT,
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(gint64 *) & value)) {
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dec->granulepos = value;
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GST_DEBUG_OBJECT (dec,
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"setting granuleposition to %" G_GUINT64_FORMAT " after discont",
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value);
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} else {
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GST_WARNING_OBJECT (dec,
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"discont event didn't include offset, we might set it wrong now");
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}
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if (dec->packetno < 3) {
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if (dec->granulepos != 0)
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
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("can't handle discont before parsing first 3 packets"));
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dec->packetno = 0;
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gst_pad_push (dec->srcpad, GST_DATA (gst_event_new_discontinuous (FALSE,
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GST_FORMAT_TIME, (guint64) 0, GST_FORMAT_DEFAULT,
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(guint64) 0, GST_FORMAT_BYTES, (guint64) 0, 0)));
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} else {
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GstFormat time_format, default_format, bytes_format;
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time_format = GST_FORMAT_TIME;
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default_format = GST_FORMAT_DEFAULT;
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bytes_format = GST_FORMAT_BYTES;
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dec->packetno = 3;
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/* if one of them works, all of them work */
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if (vorbis_dec_convert (dec->srcpad, GST_FORMAT_DEFAULT,
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dec->granulepos, &time_format, &time)
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&& vorbis_dec_convert (dec->srcpad, GST_FORMAT_DEFAULT,
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dec->granulepos, &bytes_format, &bytes)) {
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gst_pad_push (dec->srcpad,
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GST_DATA (gst_event_new_discontinuous (FALSE, GST_FORMAT_TIME,
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time, GST_FORMAT_DEFAULT, dec->granulepos,
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GST_FORMAT_BYTES, bytes, 0)));
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} else {
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GST_ERROR_OBJECT (dec,
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"failed to parse data for DISCONT event, not sending any");
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}
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#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
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vorbis_synthesis_restart (&dec->vd);
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#endif
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}
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gst_data_unref (GST_DATA (event));
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break;
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default:
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gst_pad_event_default (dec->sinkpad, event);
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break;
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}
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}
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static void
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vorbis_dec_chain (GstPad * pad, GstData * data)
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{
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GstBuffer *buf;
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GstVorbisDec *vd;
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ogg_packet packet; /* lol */
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vd = GST_VORBIS_DEC (gst_pad_get_parent (pad));
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if (GST_IS_EVENT (data)) {
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vorbis_dec_event (vd, GST_EVENT (data));
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return;
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}
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buf = GST_BUFFER (data);
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/* make ogg_packet out of the buffer */
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packet.packet = GST_BUFFER_DATA (buf);
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packet.bytes = GST_BUFFER_SIZE (buf);
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packet.granulepos = GST_BUFFER_OFFSET_END (buf);
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packet.packetno = vd->packetno++;
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if (GST_BUFFER_OFFSET_END_IS_VALID (buf))
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vd->granulepos = GST_BUFFER_OFFSET_END (buf);;
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/*
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* FIXME. Is there anyway to know that this is the last packet and
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* set e_o_s??
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*/
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packet.e_o_s = 0;
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/* switch depending on packet type */
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if (packet.packet[0] & 1) {
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/* header packet */
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if (packet.packet[0] / 2 != packet.packetno) {
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/* FIXME: just skip? */
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GST_WARNING_OBJECT (GST_ELEMENT (vd),
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"unexpected packet type %d, expected %d",
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(gint) packet.packet[0], (gint) packet.packetno);
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gst_data_unref (data);
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return;
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}
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/* Packetno = 0 if the first byte is exactly 0x01 */
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packet.b_o_s = (packet.packet[0] == 0x1) ? 1 : 0;
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if (vorbis_synthesis_headerin (&vd->vi, &vd->vc, &packet)) {
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GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
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(NULL), ("couldn't read header packet"));
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gst_data_unref (data);
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return;
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}
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if (packet.packetno == 1) {
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gchar *encoder = NULL;
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GstTagList *list =
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gst_tag_list_from_vorbiscomment_buffer (buf, "\003vorbis", 7,
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&encoder);
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if (!list) {
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GST_ERROR_OBJECT (vd, "couldn't decode comments");
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list = gst_tag_list_new ();
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}
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if (encoder) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_ENCODER, encoder, NULL);
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g_free (encoder);
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}
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_ENCODER_VERSION, vd->vi.version,
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GST_TAG_AUDIO_CODEC, "Vorbis", NULL);
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if (vd->vi.bitrate_upper > 0)
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, NULL);
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if (vd->vi.bitrate_nominal > 0)
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_NOMINAL_BITRATE, (guint) vd->vi.bitrate_nominal, NULL);
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if (vd->vi.bitrate_lower > 0)
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL);
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gst_element_found_tags_for_pad (GST_ELEMENT (vd), vd->srcpad, 0, list);
|
|
} else if (packet.packetno == 2) {
|
|
GstCaps *caps;
|
|
const GstAudioChannelPosition *pos = NULL;
|
|
|
|
/* done */
|
|
vorbis_synthesis_init (&vd->vd, &vd->vi);
|
|
vorbis_block_init (&vd->vd, &vd->vb);
|
|
caps = gst_caps_new_simple ("audio/x-raw-float",
|
|
"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, "buffer-frames", G_TYPE_INT, 0, NULL);
|
|
switch (vd->vi.channels) {
|
|
case 1:
|
|
case 2:
|
|
/* nothing */
|
|
break;
|
|
case 3:{
|
|
static 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 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 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 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:
|
|
gst_data_unref (data);
|
|
gst_caps_free (caps);
|
|
GST_ELEMENT_ERROR (vd, STREAM, NOT_IMPLEMENTED, (NULL),
|
|
("Unsupported channel count %d", vd->vi.channels));
|
|
return;
|
|
}
|
|
if (pos) {
|
|
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
|
|
}
|
|
if (!gst_pad_set_explicit_caps (vd->srcpad, caps)) {
|
|
gst_caps_free (caps);
|
|
return;
|
|
}
|
|
gst_caps_free (caps);
|
|
}
|
|
} else {
|
|
float **pcm;
|
|
guint sample_count;
|
|
|
|
if (packet.packetno < 3) {
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("no header sent yet (packet no is %d)", packet.packetno));
|
|
gst_data_unref (data);
|
|
return;
|
|
}
|
|
/* normal data packet */
|
|
if (vorbis_synthesis (&vd->vb, &packet)) {
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("couldn't read data packet"));
|
|
gst_data_unref (data);
|
|
return;
|
|
}
|
|
if (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0) {
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
|
|
(NULL), ("vorbis decoder did not accept data packet"));
|
|
gst_data_unref (data);
|
|
return;
|
|
}
|
|
sample_count = vorbis_synthesis_pcmout (&vd->vd, &pcm);
|
|
if (sample_count > 0) {
|
|
int i, j;
|
|
GstBuffer *out = gst_pad_alloc_buffer (vd->srcpad, GST_BUFFER_OFFSET_NONE,
|
|
sample_count * vd->vi.channels * sizeof (float));
|
|
float *out_data = (float *) GST_BUFFER_DATA (out);
|
|
|
|
#ifdef GST_VORBIS_DEC_SEQUENTIAL
|
|
for (i = 0; i < vd->vi.channels; i++) {
|
|
memcpy (out_data, pcm[i], sample_count * sizeof (float));
|
|
out_data += sample_count;
|
|
}
|
|
#else
|
|
for (j = 0; j < sample_count; j++) {
|
|
for (i = 0; i < vd->vi.channels; i++) {
|
|
*out_data = pcm[i][j];
|
|
out_data++;
|
|
}
|
|
}
|
|
#endif
|
|
GST_BUFFER_OFFSET (out) = vd->granulepos;
|
|
GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count;
|
|
GST_BUFFER_TIMESTAMP (out) = vd->granulepos * GST_SECOND / vd->vi.rate;
|
|
GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate;
|
|
gst_pad_push (vd->srcpad, GST_DATA (out));
|
|
vorbis_synthesis_read (&vd->vd, sample_count);
|
|
vd->granulepos += sample_count;
|
|
}
|
|
}
|
|
gst_data_unref (data);
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
vorbis_dec_change_state (GstElement * element)
|
|
{
|
|
GstVorbisDec *vd = GST_VORBIS_DEC (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
vorbis_info_init (&vd->vi);
|
|
vorbis_comment_init (&vd->vc);
|
|
break;
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
vorbis_block_clear (&vd->vb);
|
|
vorbis_dsp_clear (&vd->vd);
|
|
vorbis_comment_clear (&vd->vc);
|
|
vorbis_info_clear (&vd->vi);
|
|
vd->packetno = 0;
|
|
vd->granulepos = 0;
|
|
break;
|
|
case GST_STATE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return parent_class->change_state (element);
|
|
}
|