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be0fdf972b
Original commit message from CVS: 2005-12-05 Andy Wingo <wingo@pobox.com> * ext/a52dec/gsta52dec.c: (gst_a52dec_push): * ext/mad/gstmad.c: (gst_mad_chain): * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_alloc_buffer): * gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_chain_raw): * gst/iec958/ac3iec.c: (ac3iec_chain_raw): * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream), (gst_rmdemux_parse_packet): Update for alloc_buffer changes.
745 lines
20 KiB
C
745 lines
20 KiB
C
/* GStreamer
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* Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.net>
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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 <stdlib.h>
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#include "_stdint.h"
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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#include <a52dec/a52.h>
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#include <a52dec/mm_accel.h>
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#include "gsta52dec.h"
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#include <liboil/liboil.h>
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#include <liboil/liboilcpu.h>
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#include <liboil/liboilfunction.h>
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/* elementfactory information */
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static GstElementDetails gst_a52dec_details = {
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"ATSC A/52 audio decoder",
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"Codec/Decoder/Audio",
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"Decodes ATSC A/52 encoded audio streams",
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"David I. Lehn <dlehn@users.sourceforge.net>",
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};
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#ifdef LIBA52_DOUBLE
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#define SAMPLE_WIDTH 64
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#else
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#define SAMPLE_WIDTH 32
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#endif
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GST_DEBUG_CATEGORY_STATIC (a52dec_debug);
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#define GST_CAT_DEFAULT (a52dec_debug)
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/* A52Dec args */
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enum
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{
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ARG_0,
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ARG_DRC
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};
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static GstStaticPadTemplate sink_factory = 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-ac3; audio/ac3; audio/x-private1-ac3")
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);
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static GstStaticPadTemplate src_factory = 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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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
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"width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", "
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"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
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);
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static void gst_a52dec_base_init (GstA52DecClass * klass);
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static void gst_a52dec_class_init (GstA52DecClass * klass);
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static void gst_a52dec_init (GstA52Dec * a52dec);
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static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer);
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static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf);
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static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_a52dec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_a52dec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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GType
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gst_a52dec_get_type (void)
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{
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static GType a52dec_type = 0;
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if (!a52dec_type) {
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static const GTypeInfo a52dec_info = {
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sizeof (GstA52DecClass),
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(GBaseInitFunc) gst_a52dec_base_init,
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NULL,
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(GClassInitFunc) gst_a52dec_class_init,
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NULL,
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NULL,
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sizeof (GstA52Dec),
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0,
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(GInstanceInitFunc) gst_a52dec_init,
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};
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a52dec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstA52Dec", &a52dec_info, 0);
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}
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return a52dec_type;
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}
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static void
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gst_a52dec_base_init (GstA52DecClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_set_details (element_class, &gst_a52dec_details);
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GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0,
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"AC3/A52 software decoder");
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}
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static void
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gst_a52dec_class_init (GstA52DecClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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guint cpuflags;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_a52dec_set_property;
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gobject_class->get_property = gst_a52dec_get_property;
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
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g_param_spec_boolean ("drc", "Dynamic Range Compression",
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"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
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oil_init ();
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klass->a52_cpuflags = 0;
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cpuflags = oil_cpu_get_flags ();
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if (cpuflags & OIL_IMPL_FLAG_MMX)
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klass->a52_cpuflags |= MM_ACCEL_X86_MMX;
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if (cpuflags & OIL_IMPL_FLAG_3DNOW)
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klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW;
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if (cpuflags & OIL_IMPL_FLAG_MMXEXT)
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klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT;
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GST_LOG ("CPU flags: a52=%08x, liboil=%08x", klass->a52_cpuflags, cpuflags);
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}
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static void
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gst_a52dec_init (GstA52Dec * a52dec)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (a52dec);
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/* create the sink and src pads */
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a52dec->sinkpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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gst_pad_set_setcaps_function (a52dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps));
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gst_pad_set_chain_function (a52dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_a52dec_chain));
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gst_pad_set_event_function (a52dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
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gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
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a52dec->srcpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"src"), "src");
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gst_pad_use_fixed_caps (a52dec->srcpad);
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gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
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a52dec->dynamic_range_compression = FALSE;
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a52dec->cache = NULL;
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}
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static int
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gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
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{
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int chans = 0;
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GstAudioChannelPosition *pos = NULL;
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/* allocated just for safety. Number makes no sense */
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if (_pos) {
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pos = g_new (GstAudioChannelPosition, 6);
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*_pos = pos;
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}
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if (flags & A52_LFE) {
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chans += 1;
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if (pos) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE;
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}
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}
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flags &= A52_CHANNEL_MASK;
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switch (flags) {
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case A52_3F2R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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chans += 5;
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break;
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case A52_2F2R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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}
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chans += 4;
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break;
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case A52_3F1R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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}
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chans += 4;
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break;
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case A52_2F1R:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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}
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chans += 3;
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break;
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case A52_3F:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
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chans += 3;
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break;
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/*case A52_CHANNEL: */
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case A52_STEREO:
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case A52_DOLBY:
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if (pos) {
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pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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}
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chans += 2;
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break;
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default:
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/* error */
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g_warning ("a52dec invalid flags %d", flags);
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g_free (pos);
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return 0;
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}
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return chans;
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}
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static GstFlowReturn
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gst_a52dec_push (GstA52Dec * a52dec,
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GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
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{
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GstBuffer *buf;
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int chans, n, c;
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GstFlowReturn result;
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flags &= (A52_CHANNEL_MASK | A52_LFE);
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chans = gst_a52dec_channels (flags, NULL);
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if (!chans) {
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return GST_FLOW_ERROR;
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}
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result =
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gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
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256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
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if (result != GST_FLOW_OK)
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return result;
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for (n = 0; n < 256; n++) {
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for (c = 0; c < chans; c++) {
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((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
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samples[c * 256 + n];
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}
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}
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GST_BUFFER_TIMESTAMP (buf) = timestamp;
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GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
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GST_DEBUG_OBJECT (a52dec,
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"Pushing buffer with ts %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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return gst_pad_push (srcpad, buf);
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}
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static gboolean
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gst_a52dec_reneg (GstPad * pad)
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{
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GstAudioChannelPosition *pos;
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GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
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gint channels = gst_a52dec_channels (a52dec->using_channels, &pos);
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GstCaps *caps = NULL;
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gboolean result = FALSE;
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if (!channels)
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goto done;
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GST_INFO ("a52dec: reneg channels:%d rate:%d\n",
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channels, a52dec->sample_rate);
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caps = gst_caps_new_simple ("audio/x-raw-float",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, SAMPLE_WIDTH,
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"channels", G_TYPE_INT, channels,
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"rate", G_TYPE_INT, a52dec->sample_rate, NULL);
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gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
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g_free (pos);
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if (!gst_pad_set_caps (pad, caps))
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goto done;
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result = TRUE;
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done:
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if (caps)
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gst_caps_unref (caps);
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gst_object_unref (GST_OBJECT (a52dec));
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return result;
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}
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static gboolean
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gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
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{
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GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
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gboolean ret = FALSE;
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GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:{
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GstFormat format;
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gint64 val;
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gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL,
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NULL);
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if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) {
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GST_WARNING ("No time in newsegment event %p", event);
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} else {
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a52dec->time = val;
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}
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if (a52dec->cache) {
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gst_buffer_unref (a52dec->cache);
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a52dec->cache = NULL;
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}
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ret = gst_pad_event_default (pad, event);
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break;
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}
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case GST_EVENT_TAG:
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case GST_EVENT_EOS:{
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ret = gst_pad_event_default (pad, event);
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break;
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}
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case GST_EVENT_FLUSH_START:
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ret = gst_pad_event_default (pad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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if (a52dec->cache) {
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gst_buffer_unref (a52dec->cache);
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a52dec->cache = NULL;
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}
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ret = gst_pad_event_default (pad, event);
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break;
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default:
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ret = gst_pad_event_default (pad, event);
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break;
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}
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gst_object_unref (a52dec);
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return ret;
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}
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static void
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gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
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{
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GstTagList *taglist;
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taglist = gst_tag_list_new ();
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gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
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GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
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gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
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GST_PAD (a52dec->srcpad), taglist);
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}
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static GstFlowReturn
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gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
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guint length, gint flags, gint sample_rate, gint bit_rate)
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{
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gint channels, i;
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gboolean need_reneg = FALSE;
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/* update stream information, renegotiate or re-streaminfo if needed */
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need_reneg = FALSE;
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if (a52dec->sample_rate != sample_rate) {
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need_reneg = TRUE;
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a52dec->sample_rate = sample_rate;
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}
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if (flags) {
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a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
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}
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if (bit_rate != a52dec->bit_rate) {
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a52dec->bit_rate = bit_rate;
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gst_a52dec_update_streaminfo (a52dec);
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}
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/* process */
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flags = a52dec->request_channels; /* | A52_ADJUST_LEVEL; */
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a52dec->level = 1;
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if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
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GST_WARNING ("a52_frame error");
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return GST_FLOW_OK;
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}
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channels = flags & (A52_CHANNEL_MASK | A52_LFE);
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if (a52dec->using_channels != channels) {
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need_reneg = TRUE;
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a52dec->using_channels = channels;
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}
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/* negotiate if required */
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if (need_reneg == TRUE) {
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GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d\n",
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|
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
|
|
if (!gst_a52dec_reneg (a52dec->srcpad)) {
|
|
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
if (a52dec->dynamic_range_compression == FALSE) {
|
|
a52_dynrng (a52dec->state, NULL, NULL);
|
|
}
|
|
|
|
/* each frame consists of 6 blocks */
|
|
for (i = 0; i < 6; i++) {
|
|
if (a52_block (a52dec->state)) {
|
|
GST_WARNING ("a52_block error %d", i);
|
|
} else {
|
|
GstFlowReturn ret;
|
|
|
|
/* push on */
|
|
ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
|
|
a52dec->samples, a52dec->time);
|
|
if (ret != GST_FLOW_OK)
|
|
return ret;
|
|
}
|
|
a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
|
|
GstStructure *structure;
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
|
|
a52dec->dvdmode = TRUE;
|
|
else
|
|
a52dec->dvdmode = FALSE;
|
|
|
|
gst_object_unref (a52dec);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
|
|
GstFlowReturn ret;
|
|
|
|
if (a52dec->dvdmode) {
|
|
gint size = GST_BUFFER_SIZE (buf);
|
|
guchar *data = GST_BUFFER_DATA (buf);
|
|
gint first_access;
|
|
gint offset;
|
|
gint len;
|
|
GstBuffer *subbuf;
|
|
|
|
if (size < 2) {
|
|
GST_ERROR_OBJECT (pad, "Insufficient data in buffer. "
|
|
"Can't determine first_acess");
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
first_access = (data[0] << 8) | data[1];
|
|
|
|
/* Skip the first_access header */
|
|
offset = 2;
|
|
|
|
if (first_access > 1) {
|
|
/* Length of data before first_access */
|
|
len = first_access - 1;
|
|
|
|
if (len <= 0 || offset + len > size) {
|
|
GST_ERROR_OBJECT (pad, "Bad first_access parameter in buffer");
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
subbuf = gst_buffer_create_sub (buf, offset, len);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
|
|
ret = gst_a52dec_chain_raw (pad, subbuf);
|
|
if (ret != GST_FLOW_OK)
|
|
goto done;
|
|
|
|
offset += len;
|
|
len = size - offset;
|
|
|
|
if (len > 0) {
|
|
subbuf = gst_buffer_create_sub (buf, offset, len);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
ret = gst_a52dec_chain_raw (pad, subbuf);
|
|
}
|
|
} else {
|
|
/* No first_access, so no timestamp */
|
|
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
|
|
ret = gst_a52dec_chain_raw (pad, subbuf);
|
|
}
|
|
} else {
|
|
ret = gst_a52dec_chain_raw (pad, buf);
|
|
}
|
|
|
|
done:
|
|
gst_object_unref (a52dec);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
|
|
guint8 *data;
|
|
guint size;
|
|
gint length = 0, flags, sample_rate, bit_rate;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
|
|
/* merge with cache, if any. Also make sure timestamps match */
|
|
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
|
|
a52dec->time = GST_BUFFER_TIMESTAMP (buf);
|
|
GST_DEBUG_OBJECT (a52dec,
|
|
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
}
|
|
|
|
if (a52dec->cache) {
|
|
buf = gst_buffer_join (a52dec->cache, buf);
|
|
a52dec->cache = NULL;
|
|
}
|
|
data = GST_BUFFER_DATA (buf);
|
|
size = GST_BUFFER_SIZE (buf);
|
|
|
|
/* find and read header */
|
|
bit_rate = a52dec->bit_rate;
|
|
sample_rate = a52dec->sample_rate;
|
|
flags = 0;
|
|
while (size >= 7) {
|
|
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
|
|
if (length == 0) {
|
|
/* no sync */
|
|
data++;
|
|
size--;
|
|
} else if (length <= size) {
|
|
GST_DEBUG ("Sync: %d", length);
|
|
result = gst_a52dec_handle_frame (a52dec, data,
|
|
length, flags, sample_rate, bit_rate);
|
|
if (result != GST_FLOW_OK) {
|
|
size = 0;
|
|
break;
|
|
}
|
|
size -= length;
|
|
data += length;
|
|
} else {
|
|
/* not enough data */
|
|
GST_LOG ("Not enough data available");
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* keep cache */
|
|
if (length == 0) {
|
|
GST_LOG ("No sync found");
|
|
}
|
|
|
|
if (size > 0) {
|
|
a52dec->cache = gst_buffer_create_sub (buf,
|
|
GST_BUFFER_SIZE (buf) - size, size);
|
|
}
|
|
|
|
gst_buffer_unref (buf);
|
|
gst_object_unref (a52dec);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstA52Dec *a52dec = GST_A52DEC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:{
|
|
GstA52DecClass *klass;
|
|
|
|
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
|
|
a52dec->state = a52_init (klass->a52_cpuflags);
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
a52dec->samples = a52_samples (a52dec->state);
|
|
a52dec->bit_rate = -1;
|
|
a52dec->sample_rate = -1;
|
|
a52dec->stream_channels = A52_CHANNEL;
|
|
a52dec->request_channels = A52_3F2R | A52_LFE;
|
|
a52dec->using_channels = A52_CHANNEL;
|
|
a52dec->level = 1;
|
|
a52dec->bias = 0;
|
|
a52dec->time = 0;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
a52dec->samples = NULL;
|
|
if (a52dec->cache) {
|
|
gst_buffer_unref (a52dec->cache);
|
|
a52dec->cache = NULL;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
a52_free (a52dec->state);
|
|
a52dec->state = NULL;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstA52Dec *src = GST_A52DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DRC:
|
|
GST_OBJECT_LOCK (src);
|
|
src->dynamic_range_compression = g_value_get_boolean (value);
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstA52Dec *src = GST_A52DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DRC:
|
|
GST_OBJECT_LOCK (src);
|
|
g_value_set_boolean (value, src->dynamic_range_compression);
|
|
GST_OBJECT_UNLOCK (src);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
if (!gst_element_register (plugin, "a52dec", GST_RANK_PRIMARY,
|
|
GST_TYPE_A52DEC))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"a52dec",
|
|
"Decodes ATSC A/52 encoded audio streams",
|
|
plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN);
|