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144 lines
4.5 KiB
C
144 lines
4.5 KiB
C
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/* GStreamer
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* Copyright (C) <2009> Jan Schmidt <thaytan@noraisin.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 "rsnaudiodec.h"
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GST_DEBUG_CATEGORY_STATIC (rsn_audiodec_debug);
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#define GST_CAT_DEFAULT rsn_audiodec_debug
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static GstStaticPadTemplate sink_template = 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/mpeg,mpegversion=(int)1;"
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"audio/x-private1-lpcm;"
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"audio/x-private1-ac3;" "audio/x-private1-dts;" "audio/ac3")
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);
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static GstStaticPadTemplate src_template = 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) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) { 32, 64 }; "
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"audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, "
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"depth = (int) 32, "
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"signed = (boolean) true; "
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"audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 24, "
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"depth = (int) 24, "
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"signed = (boolean) true; "
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"audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"signed = (boolean) true; "
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"audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 8, " "depth = (int) 8, " "signed = (boolean) true")
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);
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G_DEFINE_TYPE (RsnAudioDec, rsn_audiodec, GST_TYPE_BIN);
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static gboolean rsn_audiodec_set_sink_caps (GstPad * sinkpad, GstCaps * caps);
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static GstCaps *rsn_audiodec_get_sink_caps (GstPad * sinkpad);
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static GstFlowReturn rsn_audiodec_chain (GstPad * pad, GstBuffer * buf);
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static gboolean rsn_audiodec_sink_event (GstPad * pad, GstEvent * event);
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static void
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rsn_audiodec_class_init (RsnAudioDecClass * klass)
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{
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static GstElementDetails element_details = {
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"RsnAudioDec",
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"Audio/Decoder",
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"Resin DVD audio stream decoder",
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"Jan Schmidt <thaytan@noraisin.net>"
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};
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (rsn_audiodec_debug, "rsn_audiodec",
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0, "Resin DVD audio stream decoder");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_details (element_class, &element_details);
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}
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static void
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rsn_audiodec_init (RsnAudioDec * self)
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{
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self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_setcaps_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (rsn_audiodec_set_sink_caps));
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gst_pad_set_getcaps_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (rsn_audiodec_get_sink_caps));
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gst_pad_set_chain_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (rsn_audiodec_chain));
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gst_pad_set_event_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (rsn_audiodec_sink_event));
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gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
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self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
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}
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static gboolean
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rsn_audiodec_set_sink_caps (GstPad * sinkpad, GstCaps * caps)
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{
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return FALSE;
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}
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static GstCaps *
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rsn_audiodec_get_sink_caps (GstPad * sinkpad)
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{
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return NULL;
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}
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static GstFlowReturn
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rsn_audiodec_chain (GstPad * pad, GstBuffer * buf)
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{
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return GST_FLOW_ERROR;
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}
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static gboolean
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rsn_audiodec_sink_event (GstPad * pad, GstEvent * event)
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{
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return FALSE;
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}
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