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1349 lines
37 KiB
C
1349 lines
37 KiB
C
/* GStreamer FAAD (Free AAC Decoder) plugin
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* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular 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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/**
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* SECTION:element-faad
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* @seealso: faac
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*
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* faad decodes AAC (MPEG-4 part 3) stream.
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*
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* <refsect2>
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* <title>Example launch lines</title>
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* |[
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* gst-launch filesrc location=example.mp4 ! qtdemux ! faad ! audioconvert ! audioresample ! autoaudiosink
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* ]| Play aac from mp4 file.
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* |[
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* gst-launch filesrc location=example.adts ! faad ! audioconvert ! audioresample ! autoaudiosink
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* ]| Play standalone aac bitstream.
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/multichannel.h>
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/* These are the correct types for these functions, as defined in the source,
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* with types changed to match glib types, since those are defined for us.
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* However, upstream FAAD is distributed with a broken header file that defined
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* these wrongly (in a way which was broken on 64 bit systems).
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*
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* Upstream CVS still has the bug, but has also renamed all the public symbols
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* for Better Corporate Branding (or whatever), so we need to take that
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* (FAAD_IS_NEAAC) into account as well.
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*
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* We must call them using these definitions. Most distributions now have the
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* corrected header file (they distribute a patch along with the source),
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* but not all, hence this Truly Evil Hack.
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*
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* Note: The prototypes don't need to be defined conditionaly, as the cpp will
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* do that for us.
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*/
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#if FAAD2_MINOR_VERSION < 7
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#ifdef FAAD_IS_NEAAC
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#define NeAACDecInit NeAACDecInit_no_definition
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#define NeAACDecInit2 NeAACDecInit2_no_definition
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#else
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#define faacDecInit faacDecInit_no_definition
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#define faacDecInit2 faacDecInit2_no_definition
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#endif
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#endif /* FAAD2_MINOR_VERSION < 7 */
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#include "gstfaad.h"
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#if FAAD2_MINOR_VERSION < 7
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#ifdef FAAD_IS_NEAAC
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#undef NeAACDecInit
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#undef NeAACDecInit2
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#else
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#undef faacDecInit
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#undef faacDecInit2
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#endif
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extern long faacDecInit (faacDecHandle, guint8 *, guint32, guint32 *, guint8 *);
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extern gint8 faacDecInit2 (faacDecHandle, guint8 *, guint32,
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guint32 *, guint8 *);
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#endif /* FAAD2_MINOR_VERSION < 7 */
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GST_DEBUG_CATEGORY_STATIC (faad_debug);
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#define GST_CAT_DEFAULT faad_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) { 2, 4 }")
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);
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#define STATIC_INT_CAPS(bpp) \
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"audio/x-raw-int, " \
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"endianness = (int) BYTE_ORDER, " \
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"signed = (bool) TRUE, " \
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"width = (int) " G_STRINGIFY (bpp) ", " \
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"depth = (int) " G_STRINGIFY (bpp) ", " \
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"rate = (int) [ 8000, 96000 ], " \
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"channels = (int) [ 1, 8 ]"
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#if 0
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#define STATIC_FLOAT_CAPS(bpp) \
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"audio/x-raw-float, " \
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"endianness = (int) BYTE_ORDER, " \
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"depth = (int) " G_STRINGIFY (bpp) ", " \
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"rate = (int) [ 8000, 96000 ], " \
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"channels = (int) [ 1, 8 ]"
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#endif
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/*
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* All except 16-bit integer are disabled until someone fixes FAAD.
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* FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
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* for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp
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* audio, but not for any other. You'll get random segfaults, crashes
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* and even valgrind goes crazy.
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*/
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#define STATIC_CAPS \
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STATIC_INT_CAPS (16)
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#if 0
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#define NOTUSED "; " \
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STATIC_INT_CAPS (24) \
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"; " \
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STATIC_INT_CAPS (32) \
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"; " \
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STATIC_FLOAT_CAPS (32) \
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"; " \
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STATIC_FLOAT_CAPS (64)
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#endif
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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 (STATIC_CAPS)
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);
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static void gst_faad_base_init (GstFaadClass * klass);
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static void gst_faad_class_init (GstFaadClass * klass);
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static void gst_faad_init (GstFaad * faad);
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static void gst_faad_reset (GstFaad * faad);
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static void gst_faad_finalize (GObject * object);
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static void clear_queued (GstFaad * faad);
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static gboolean gst_faad_setcaps (GstPad * pad, GstCaps * caps);
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static gboolean gst_faad_src_event (GstPad * pad, GstEvent * event);
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static gboolean gst_faad_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_faad_src_query (GstPad * pad, GstQuery * query);
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static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer);
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static GstStateChangeReturn gst_faad_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_faad_src_convert (GstFaad * faad, GstFormat src_format,
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gint64 src_val, GstFormat dest_format, gint64 * dest_val);
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static gboolean gst_faad_open_decoder (GstFaad * faad);
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static void gst_faad_close_decoder (GstFaad * faad);
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static GstElementClass *parent_class; /* NULL */
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GType
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gst_faad_get_type (void)
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{
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static GType gst_faad_type = 0;
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if (!gst_faad_type) {
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static const GTypeInfo gst_faad_info = {
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sizeof (GstFaadClass),
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(GBaseInitFunc) gst_faad_base_init,
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NULL,
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(GClassInitFunc) gst_faad_class_init,
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NULL,
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NULL,
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sizeof (GstFaad),
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0,
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(GInstanceInitFunc) gst_faad_init,
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};
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gst_faad_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstFaad", &gst_faad_info, 0);
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}
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return gst_faad_type;
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}
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static void
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gst_faad_base_init (GstFaadClass * 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 (&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_simple (element_class, "AAC audio decoder",
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"Codec/Decoder/Audio",
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"Free MPEG-2/4 AAC decoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
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}
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static void
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gst_faad_class_init (GstFaadClass * klass)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_faad_finalize);
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faad_change_state);
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}
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static void
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gst_faad_init (GstFaad * faad)
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{
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faad->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
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gst_pad_set_event_function (faad->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faad_sink_event));
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gst_pad_set_setcaps_function (faad->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faad_setcaps));
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gst_pad_set_chain_function (faad->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faad_chain));
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faad->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (faad->srcpad);
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gst_pad_set_query_function (faad->srcpad,
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GST_DEBUG_FUNCPTR (gst_faad_src_query));
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gst_pad_set_event_function (faad->srcpad,
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GST_DEBUG_FUNCPTR (gst_faad_src_event));
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gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
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faad->adapter = gst_adapter_new ();
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gst_faad_reset (faad);
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}
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static void
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gst_faad_reset_stream_state (GstFaad * faad)
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{
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faad->sync_flush = 0;
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gst_adapter_clear (faad->adapter);
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clear_queued (faad);
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if (faad->handle)
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faacDecPostSeekReset (faad->handle, 0);
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}
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static void
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gst_faad_reset (GstFaad * faad)
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{
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gst_segment_init (&faad->segment, GST_FORMAT_TIME);
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faad->samplerate = -1;
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faad->channels = -1;
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faad->init = FALSE;
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faad->packetised = FALSE;
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g_free (faad->channel_positions);
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faad->channel_positions = NULL;
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faad->next_ts = 0;
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faad->prev_ts = GST_CLOCK_TIME_NONE;
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faad->bytes_in = 0;
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faad->sum_dur_out = 0;
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faad->error_count = 0;
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gst_faad_reset_stream_state (faad);
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}
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static void
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gst_faad_finalize (GObject * object)
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{
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GstFaad *faad = GST_FAAD (object);
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g_object_unref (faad->adapter);
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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_faad_send_tags (GstFaad * faad)
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{
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GstTagList *tags;
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tags = gst_tag_list_new ();
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gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
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GST_TAG_AUDIO_CODEC, "MPEG-4 AAC audio", NULL);
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gst_element_found_tags (GST_ELEMENT (faad), tags);
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}
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static gint
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aac_rate_idx (gint rate)
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{
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if (92017 <= rate)
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return 0;
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else if (75132 <= rate)
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return 1;
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else if (55426 <= rate)
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return 2;
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else if (46009 <= rate)
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return 3;
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else if (37566 <= rate)
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return 4;
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else if (27713 <= rate)
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return 5;
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else if (23004 <= rate)
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return 6;
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else if (18783 <= rate)
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return 7;
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else if (13856 <= rate)
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return 8;
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else if (11502 <= rate)
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return 9;
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else if (9391 <= rate)
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return 10;
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else
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return 11;
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}
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static gboolean
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gst_faad_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
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GstStructure *str = gst_caps_get_structure (caps, 0);
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GstBuffer *buf;
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const GValue *value;
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/* Assume raw stream */
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faad->packetised = FALSE;
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if ((value = gst_structure_get_value (str, "codec_data"))) {
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#if FAAD2_MINOR_VERSION >= 7
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unsigned long samplerate;
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#else
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guint32 samplerate;
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#endif
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guint8 channels;
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guint8 *cdata;
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guint csize;
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/* We have codec data, means packetised stream */
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faad->packetised = TRUE;
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buf = gst_value_get_buffer (value);
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g_return_val_if_fail (buf != NULL, FALSE);
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cdata = GST_BUFFER_DATA (buf);
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csize = GST_BUFFER_SIZE (buf);
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if (csize < 2)
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goto wrong_length;
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GST_DEBUG_OBJECT (faad,
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"codec_data: object_type=%d, sample_rate=%d, channels=%d",
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((cdata[0] & 0xf8) >> 3),
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(((cdata[0] & 0x07) << 1) | ((cdata[1] & 0x80) >> 7)),
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((cdata[1] & 0x78) >> 3));
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/* someone forgot that char can be unsigned when writing the API */
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if ((gint8) faacDecInit2 (faad->handle, cdata, csize, &samplerate,
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&channels) < 0)
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goto init_failed;
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if (channels != ((cdata[1] & 0x78) >> 3)) {
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/* https://bugs.launchpad.net/ubuntu/+source/faad2/+bug/290259 */
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GST_WARNING_OBJECT (faad,
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"buggy faad version, wrong nr of channels %d instead of %d", channels,
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((cdata[1] & 0x78) >> 3));
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}
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GST_DEBUG_OBJECT (faad, "codec_data init: channels=%u, rate=%u", channels,
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(guint32) samplerate);
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/* not updating these here, so they are updated in the
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* chain function, and new caps are created etc. */
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faad->samplerate = 0;
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faad->channels = 0;
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faad->init = TRUE;
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gst_faad_send_tags (faad);
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gst_adapter_clear (faad->adapter);
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} else if ((value = gst_structure_get_value (str, "framed")) &&
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g_value_get_boolean (value) == TRUE) {
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faad->packetised = TRUE;
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GST_DEBUG_OBJECT (faad, "we have packetized audio");
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} else {
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faad->init = FALSE;
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}
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faad->fake_codec_data[0] = 0;
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faad->fake_codec_data[1] = 0;
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if (faad->packetised && !faad->init) {
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gint rate, channels;
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if (gst_structure_get_int (str, "rate", &rate) &&
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gst_structure_get_int (str, "channels", &channels)) {
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gint rate_idx, profile;
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profile = 3; /* 0=MAIN, 1=LC, 2=SSR, 3=LTP */
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rate_idx = aac_rate_idx (rate);
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faad->fake_codec_data[0] = ((profile + 1) << 3) | ((rate_idx & 0xE) >> 1);
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faad->fake_codec_data[1] = ((rate_idx & 0x1) << 7) | (channels << 3);
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GST_LOG_OBJECT (faad, "created fake codec data (%u,%u): 0x%x 0x%x", rate,
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channels, (int) faad->fake_codec_data[0],
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(int) faad->fake_codec_data[1]);
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}
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}
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gst_object_unref (faad);
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return TRUE;
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/* ERRORS */
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wrong_length:
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{
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GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
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gst_object_unref (faad);
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return FALSE;
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}
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init_failed:
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{
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GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
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gst_object_unref (faad);
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return FALSE;
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}
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}
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static GstAudioChannelPosition *
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gst_faad_chanpos_to_gst (GstFaad * faad, guchar * fpos, guint num,
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gboolean * channel_map_failed)
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{
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GstAudioChannelPosition *pos;
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guint n;
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gboolean unknown_channel = FALSE;
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*channel_map_failed = FALSE;
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/* special handling for the common cases for mono and stereo */
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if (num == 1 && fpos[0] == FRONT_CHANNEL_CENTER) {
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GST_DEBUG_OBJECT (faad, "mono common case; won't set channel positions");
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return NULL;
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} else if (num == 2 && fpos[0] == FRONT_CHANNEL_LEFT
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&& fpos[1] == FRONT_CHANNEL_RIGHT) {
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GST_DEBUG_OBJECT (faad, "stereo common case; won't set channel positions");
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return NULL;
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}
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|
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pos = g_new (GstAudioChannelPosition, num);
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for (n = 0; n < num; n++) {
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GST_DEBUG_OBJECT (faad, "faad channel %d as %d", n, fpos[n]);
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switch (fpos[n]) {
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case FRONT_CHANNEL_LEFT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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break;
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case FRONT_CHANNEL_RIGHT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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break;
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case FRONT_CHANNEL_CENTER:
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/* argh, mono = center */
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if (num == 1)
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pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
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else
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
|
|
break;
|
|
case SIDE_CHANNEL_LEFT:
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
|
|
break;
|
|
case SIDE_CHANNEL_RIGHT:
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
|
|
break;
|
|
case BACK_CHANNEL_LEFT:
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
|
|
break;
|
|
case BACK_CHANNEL_RIGHT:
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
|
|
break;
|
|
case BACK_CHANNEL_CENTER:
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
|
|
break;
|
|
case LFE_CHANNEL:
|
|
pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE;
|
|
break;
|
|
default:
|
|
GST_DEBUG_OBJECT (faad, "unknown channel %d at %d", fpos[n], n);
|
|
unknown_channel = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
if (unknown_channel) {
|
|
g_free (pos);
|
|
pos = NULL;
|
|
switch (num) {
|
|
case 1:{
|
|
GST_DEBUG_OBJECT (faad,
|
|
"FAAD reports unknown 1 channel mapping. Forcing to mono");
|
|
break;
|
|
}
|
|
case 2:{
|
|
GST_DEBUG_OBJECT (faad,
|
|
"FAAD reports unknown 2 channel mapping. Forcing to stereo");
|
|
break;
|
|
}
|
|
default:{
|
|
GST_WARNING_OBJECT (faad,
|
|
"Unsupported FAAD channel position 0x%x encountered", fpos[n]);
|
|
*channel_map_failed = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
return pos;
|
|
}
|
|
|
|
static void
|
|
clear_queued (GstFaad * faad)
|
|
{
|
|
g_list_foreach (faad->queued, (GFunc) gst_mini_object_unref, NULL);
|
|
g_list_free (faad->queued);
|
|
faad->queued = NULL;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
flush_queued (GstFaad * faad)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
while (faad->queued) {
|
|
GstBuffer *buf = GST_BUFFER_CAST (faad->queued->data);
|
|
|
|
GST_LOG_OBJECT (faad, "pushing buffer %p, timestamp %"
|
|
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
|
|
/* iterate ouput queue an push downstream */
|
|
ret = gst_pad_push (faad->srcpad, buf);
|
|
|
|
faad->queued = g_list_delete_link (faad->queued, faad->queued);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_faad_drain (GstFaad * faad)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
if (faad->segment.rate < 0.0) {
|
|
/* if we have some queued frames for reverse playback, flush
|
|
* them now */
|
|
ret = flush_queued (faad);
|
|
} else {
|
|
/* squeeze any possible remaining frames that are pending sync */
|
|
gst_faad_chain (faad->sinkpad, NULL);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_faad_do_raw_seek (GstFaad * faad, GstEvent * event)
|
|
{
|
|
GstSeekFlags flags;
|
|
GstSeekType start_type, end_type;
|
|
GstFormat format;
|
|
gdouble rate;
|
|
gint64 start, start_time;
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags, &start_type,
|
|
&start_time, &end_type, NULL);
|
|
|
|
if (rate != 1.0 ||
|
|
format != GST_FORMAT_TIME ||
|
|
start_type != GST_SEEK_TYPE_SET || end_type != GST_SEEK_TYPE_NONE) {
|
|
return FALSE;
|
|
}
|
|
|
|
if (!gst_faad_src_convert (faad, GST_FORMAT_TIME, start_time,
|
|
GST_FORMAT_BYTES, &start)) {
|
|
return FALSE;
|
|
}
|
|
|
|
event = gst_event_new_seek (1.0, GST_FORMAT_BYTES, flags,
|
|
GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE, -1);
|
|
|
|
GST_DEBUG_OBJECT (faad, "seeking to %" GST_TIME_FORMAT " at byte offset %"
|
|
G_GINT64_FORMAT, GST_TIME_ARGS (start_time), start);
|
|
|
|
return gst_pad_push_event (faad->sinkpad, event);
|
|
}
|
|
|
|
static gboolean
|
|
gst_faad_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstFaad *faad;
|
|
gboolean res;
|
|
|
|
faad = GST_FAAD (gst_pad_get_parent (pad));
|
|
|
|
GST_LOG_OBJECT (faad, "Handling %s event", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:{
|
|
/* try upstream first, there might be a demuxer */
|
|
gst_event_ref (event);
|
|
if (!(res = gst_pad_push_event (faad->sinkpad, event))) {
|
|
res = gst_faad_do_raw_seek (faad, event);
|
|
}
|
|
gst_event_unref (event);
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_push_event (faad->sinkpad, event);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (faad);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_faad_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstFaad *faad;
|
|
gboolean res = TRUE;
|
|
|
|
faad = GST_FAAD (gst_pad_get_parent (pad));
|
|
|
|
GST_LOG_OBJECT (faad, "Handling %s event", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_faad_reset_stream_state (faad);
|
|
res = gst_pad_push_event (faad->srcpad, event);
|
|
break;
|
|
case GST_EVENT_EOS:
|
|
gst_faad_drain (faad);
|
|
gst_faad_reset_stream_state (faad);
|
|
res = gst_pad_push_event (faad->srcpad, event);
|
|
break;
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
GstFormat fmt;
|
|
gboolean is_update;
|
|
gint64 start, end, base;
|
|
gdouble rate;
|
|
|
|
gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
|
|
&end, &base);
|
|
|
|
/* drain queued buffers before we activate the new segment */
|
|
gst_faad_drain (faad);
|
|
|
|
if (fmt == GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (faad,
|
|
"Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
|
|
GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (end));
|
|
gst_segment_set_newsegment (&faad->segment, is_update, rate, fmt, start,
|
|
end, base);
|
|
} else if (fmt == GST_FORMAT_BYTES) {
|
|
gint64 new_start = 0;
|
|
gint64 new_end = -1;
|
|
|
|
GST_DEBUG_OBJECT (faad, "Got NEWSEGMENT event in GST_FORMAT_BYTES (%"
|
|
G_GUINT64_FORMAT " - %" G_GUINT64_FORMAT ")", start, end);
|
|
|
|
if (gst_faad_src_convert (faad, GST_FORMAT_BYTES, start,
|
|
GST_FORMAT_TIME, &new_start)) {
|
|
if (end != -1) {
|
|
gst_faad_src_convert (faad, GST_FORMAT_BYTES, end,
|
|
GST_FORMAT_TIME, &new_end);
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (faad,
|
|
"no average bitrate yet, sending newsegment with start at 0");
|
|
}
|
|
gst_event_unref (event);
|
|
|
|
event = gst_event_new_new_segment (is_update, rate,
|
|
GST_FORMAT_TIME, new_start, new_end, new_start);
|
|
|
|
gst_segment_set_newsegment (&faad->segment, is_update, rate,
|
|
GST_FORMAT_TIME, new_start, new_end, new_start);
|
|
|
|
GST_DEBUG_OBJECT (faad,
|
|
"Sending new NEWSEGMENT event, time %" GST_TIME_FORMAT
|
|
" - %" GST_TIME_FORMAT, GST_TIME_ARGS (new_start),
|
|
GST_TIME_ARGS (new_end));
|
|
|
|
faad->next_ts = new_start;
|
|
faad->prev_ts = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
res = gst_pad_push_event (faad->srcpad, event);
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (faad);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_faad_src_convert (GstFaad * faad, GstFormat src_format, gint64 src_val,
|
|
GstFormat dest_format, gint64 * dest_val)
|
|
{
|
|
guint64 bytes_in, time_out, val;
|
|
|
|
if (src_format == dest_format) {
|
|
if (dest_val)
|
|
*dest_val = src_val;
|
|
return TRUE;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (faad);
|
|
bytes_in = faad->bytes_in;
|
|
time_out = faad->sum_dur_out;
|
|
GST_OBJECT_UNLOCK (faad);
|
|
|
|
if (bytes_in == 0 || time_out == 0)
|
|
return FALSE;
|
|
|
|
/* convert based on the average bitrate so far */
|
|
if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME) {
|
|
val = gst_util_uint64_scale (src_val, time_out, bytes_in);
|
|
} else if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES) {
|
|
val = gst_util_uint64_scale (src_val, bytes_in, time_out);
|
|
} else {
|
|
return FALSE;
|
|
}
|
|
|
|
if (dest_val)
|
|
*dest_val = (gint64) val;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_faad_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstFaad *faad;
|
|
GstPad *peer = NULL;
|
|
|
|
faad = GST_FAAD (gst_pad_get_parent (pad));
|
|
|
|
GST_LOG_OBJECT (faad, "processing %s query", GST_QUERY_TYPE_NAME (query));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_DURATION:{
|
|
GstFormat format;
|
|
gint64 len_bytes, duration;
|
|
|
|
/* try upstream first, in case there's a demuxer */
|
|
if ((res = gst_pad_query_default (pad, query)))
|
|
break;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
if (format != GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (faad, "query failed: can't handle format %s",
|
|
gst_format_get_name (format));
|
|
break;
|
|
}
|
|
|
|
peer = gst_pad_get_peer (faad->sinkpad);
|
|
if (peer == NULL)
|
|
break;
|
|
|
|
format = GST_FORMAT_BYTES;
|
|
if (!gst_pad_query_duration (peer, &format, &len_bytes)) {
|
|
GST_DEBUG_OBJECT (faad, "query failed: failed to get upstream length");
|
|
break;
|
|
}
|
|
|
|
res = gst_faad_src_convert (faad, GST_FORMAT_BYTES, len_bytes,
|
|
GST_FORMAT_TIME, &duration);
|
|
|
|
if (res) {
|
|
gst_query_set_duration (query, GST_FORMAT_TIME, duration);
|
|
|
|
GST_LOG_OBJECT (faad, "duration estimate: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (duration));
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_POSITION:{
|
|
GstFormat format;
|
|
gint64 pos_bytes, pos;
|
|
|
|
/* try upstream first, in case there's a demuxer */
|
|
if ((res = gst_pad_query_default (pad, query)))
|
|
break;
|
|
|
|
gst_query_parse_position (query, &format, NULL);
|
|
if (format != GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (faad, "query failed: can't handle format %s",
|
|
gst_format_get_name (format));
|
|
break;
|
|
}
|
|
|
|
peer = gst_pad_get_peer (faad->sinkpad);
|
|
if (peer == NULL)
|
|
break;
|
|
|
|
format = GST_FORMAT_BYTES;
|
|
if (!gst_pad_query_position (peer, &format, &pos_bytes)) {
|
|
GST_OBJECT_LOCK (faad);
|
|
pos = faad->next_ts;
|
|
GST_OBJECT_UNLOCK (faad);
|
|
res = TRUE;
|
|
} else {
|
|
res = gst_faad_src_convert (faad, GST_FORMAT_BYTES, pos_bytes,
|
|
GST_FORMAT_TIME, &pos);
|
|
}
|
|
|
|
if (res) {
|
|
gst_query_set_position (query, GST_FORMAT_TIME, pos);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
|
|
if (peer)
|
|
gst_object_unref (peer);
|
|
|
|
gst_object_unref (faad);
|
|
return res;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info)
|
|
{
|
|
GstAudioChannelPosition *pos;
|
|
gboolean ret;
|
|
gboolean channel_map_failed;
|
|
GstCaps *caps;
|
|
gboolean fmt_change = FALSE;
|
|
|
|
/* see if we need to renegotiate */
|
|
if (info->samplerate != faad->samplerate ||
|
|
info->channels != faad->channels || !faad->channel_positions) {
|
|
fmt_change = TRUE;
|
|
} else {
|
|
gint i;
|
|
|
|
for (i = 0; i < info->channels; i++) {
|
|
if (info->channel_position[i] != faad->channel_positions[i])
|
|
fmt_change = TRUE;
|
|
}
|
|
}
|
|
|
|
if (G_LIKELY (!fmt_change))
|
|
return TRUE;
|
|
|
|
/* store new negotiation information */
|
|
faad->samplerate = info->samplerate;
|
|
faad->channels = info->channels;
|
|
g_free (faad->channel_positions);
|
|
faad->channel_positions = g_memdup (info->channel_position, faad->channels);
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw-int",
|
|
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
"signed", G_TYPE_BOOLEAN, TRUE,
|
|
"width", G_TYPE_INT, 16,
|
|
"depth", G_TYPE_INT, 16,
|
|
"rate", G_TYPE_INT, faad->samplerate,
|
|
"channels", G_TYPE_INT, faad->channels, NULL);
|
|
|
|
faad->bps = 16 / 8;
|
|
|
|
channel_map_failed = FALSE;
|
|
pos =
|
|
gst_faad_chanpos_to_gst (faad, faad->channel_positions, faad->channels,
|
|
&channel_map_failed);
|
|
if (channel_map_failed) {
|
|
GST_DEBUG_OBJECT (faad, "Could not map channel positions");
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
if (pos) {
|
|
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
|
|
g_free (pos);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (faad, "New output caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
ret = gst_pad_set_caps (faad->srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Find syncpoint in ADTS/ADIF stream. Doesn't work for raw,
|
|
* packetized streams. Be careful when calling.
|
|
* Returns FALSE on no-sync, fills offset/length if one/two
|
|
* syncpoints are found, only returns TRUE when it finds two
|
|
* subsequent syncpoints (similar to mp3 typefinding in
|
|
* gst/typefind/) for ADTS because 12 bits isn't very reliable.
|
|
*/
|
|
static gboolean
|
|
gst_faad_sync (GstFaad * faad, guint8 * data, guint size, gboolean next,
|
|
guint * off)
|
|
{
|
|
guint n = 0;
|
|
gint snc;
|
|
gboolean ret = FALSE;
|
|
|
|
GST_LOG_OBJECT (faad, "Finding syncpoint");
|
|
|
|
/* check for too small a buffer */
|
|
if (size < 3)
|
|
goto exit;
|
|
|
|
for (n = 0; n < size - 3; n++) {
|
|
snc = GST_READ_UINT16_BE (&data[n]);
|
|
if ((snc & 0xfff6) == 0xfff0) {
|
|
/* we have an ADTS syncpoint. Parse length and find
|
|
* next syncpoint. */
|
|
guint len;
|
|
|
|
GST_LOG_OBJECT (faad,
|
|
"Found one ADTS syncpoint at offset 0x%x, tracing next...", n);
|
|
|
|
if (size - n < 5) {
|
|
GST_LOG_OBJECT (faad, "Not enough data to parse ADTS header");
|
|
break;
|
|
}
|
|
|
|
len = ((data[n + 3] & 0x03) << 11) |
|
|
(data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5);
|
|
if (n + len + 2 >= size) {
|
|
GST_LOG_OBJECT (faad, "Frame size %d, next frame is not within reach",
|
|
len);
|
|
if (next) {
|
|
break;
|
|
} else if (n + len <= size) {
|
|
GST_LOG_OBJECT (faad, "but have complete frame and no next frame; "
|
|
"accept ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
snc = GST_READ_UINT16_BE (&data[n + len]);
|
|
if ((snc & 0xfff6) == 0xfff0) {
|
|
GST_LOG_OBJECT (faad,
|
|
"Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
|
|
GST_LOG_OBJECT (faad, "No next frame found... (should be at 0x%x)",
|
|
n + len);
|
|
} else if (!memcmp (&data[n], "ADIF", 4)) {
|
|
/* we have an ADIF syncpoint. 4 bytes is enough. */
|
|
GST_LOG_OBJECT (faad, "Found ADIF syncpoint at offset 0x%x", n);
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
exit:
|
|
*off = n;
|
|
|
|
if (!ret)
|
|
GST_LOG_OBJECT (faad, "Found no syncpoint");
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
looks_like_valid_header (guint8 * input_data, guint input_size)
|
|
{
|
|
if (input_size < 4)
|
|
return FALSE;
|
|
|
|
if (input_data[0] == 'A'
|
|
&& input_data[1] == 'D' && input_data[2] == 'I' && input_data[3] == 'F')
|
|
/* ADIF type header */
|
|
return TRUE;
|
|
|
|
if (input_data[0] == 0xff && (input_data[1] >> 4) == 0xf)
|
|
/* ADTS type header */
|
|
return TRUE;
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
#define FAAD_MAX_ERROR 10
|
|
#define FAAD_MAX_SYNC 10 * 8 * 1024
|
|
|
|
static GstFlowReturn
|
|
gst_faad_chain (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
guint input_size;
|
|
guint available;
|
|
guchar *input_data;
|
|
GstFaad *faad;
|
|
GstBuffer *outbuf;
|
|
faacDecFrameInfo info;
|
|
void *out;
|
|
gboolean run_loop = TRUE;
|
|
guint sync_off;
|
|
GstClockTime ts;
|
|
gboolean next;
|
|
|
|
faad = GST_FAAD (gst_pad_get_parent (pad));
|
|
|
|
if (G_LIKELY (buffer)) {
|
|
GST_LOG_OBJECT (faad, "buffer of size %d with ts: %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
|
|
|
|
if (GST_BUFFER_IS_DISCONT (buffer)) {
|
|
gst_faad_drain (faad);
|
|
gst_faad_reset_stream_state (faad);
|
|
faad->discont = TRUE;
|
|
}
|
|
|
|
gst_adapter_push (faad->adapter, buffer);
|
|
buffer = NULL;
|
|
next = TRUE;
|
|
} else {
|
|
next = FALSE;
|
|
}
|
|
|
|
ts = gst_adapter_prev_timestamp (faad->adapter, NULL);
|
|
if (GST_CLOCK_TIME_IS_VALID (ts) && (ts != faad->prev_ts))
|
|
faad->prev_ts = faad->next_ts = ts;
|
|
|
|
available = gst_adapter_available (faad->adapter);
|
|
input_size = available;
|
|
|
|
if (G_UNLIKELY (!available))
|
|
goto out;
|
|
|
|
input_data = (guchar *) gst_adapter_peek (faad->adapter, available);
|
|
|
|
if (!faad->packetised) {
|
|
if (!gst_faad_sync (faad, input_data, input_size, next, &sync_off)) {
|
|
faad->sync_flush += sync_off;
|
|
input_size -= sync_off;
|
|
if (faad->sync_flush > FAAD_MAX_SYNC)
|
|
goto parse_failed;
|
|
else
|
|
goto out;
|
|
} else {
|
|
faad->sync_flush = 0;
|
|
input_data += sync_off;
|
|
input_size -= sync_off;
|
|
}
|
|
}
|
|
|
|
/* init if not already done during capsnego */
|
|
if (!faad->init) {
|
|
#if FAAD2_MINOR_VERSION >= 7
|
|
unsigned long rate;
|
|
#else
|
|
guint32 rate;
|
|
#endif
|
|
guint8 ch;
|
|
|
|
GST_DEBUG_OBJECT (faad, "initialising ...");
|
|
/* We check if the first data looks like it might plausibly contain
|
|
* appropriate initialisation info... if not, we use our fake_codec_data
|
|
*/
|
|
if (looks_like_valid_header (input_data, input_size) || !faad->packetised) {
|
|
if (faacDecInit (faad->handle, input_data, input_size, &rate, &ch) < 0)
|
|
goto init_failed;
|
|
|
|
GST_DEBUG_OBJECT (faad, "faacDecInit() ok: rate=%u,channels=%u",
|
|
(guint32) rate, ch);
|
|
} else {
|
|
if ((gint8) faacDecInit2 (faad->handle, faad->fake_codec_data, 2,
|
|
&rate, &ch) < 0) {
|
|
goto init2_failed;
|
|
}
|
|
GST_DEBUG_OBJECT (faad, "faacDecInit2() ok: rate=%u,channels=%u",
|
|
(guint32) rate, ch);
|
|
}
|
|
|
|
faad->init = TRUE;
|
|
gst_faad_send_tags (faad);
|
|
|
|
/* make sure we create new caps below */
|
|
faad->samplerate = 0;
|
|
faad->channels = 0;
|
|
}
|
|
|
|
/* decode cycle */
|
|
info.bytesconsumed = input_size;
|
|
info.error = 0;
|
|
|
|
while ((input_size > 0) && run_loop) {
|
|
|
|
if (faad->packetised) {
|
|
/* Only one packet per buffer, no matter how much is really consumed */
|
|
run_loop = FALSE;
|
|
} else {
|
|
if (input_size < FAAD_MIN_STREAMSIZE || info.bytesconsumed <= 0) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
out = faacDecDecode (faad->handle, &info, input_data, input_size);
|
|
|
|
if (info.error > 0) {
|
|
/* mark discont for the next buffer */
|
|
faad->discont = TRUE;
|
|
/* flush a bit, arranges for resync next time */
|
|
input_size--;
|
|
faad->error_count++;
|
|
/* do not bail out at once, but know when to stop */
|
|
if (faad->error_count > FAAD_MAX_ERROR)
|
|
goto decode_failed;
|
|
else {
|
|
GST_WARNING_OBJECT (faad, "decoding error: %s",
|
|
faacDecGetErrorMessage (info.error));
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
/* ok again */
|
|
faad->error_count = 0;
|
|
|
|
GST_LOG_OBJECT (faad, "%d bytes consumed, %d samples decoded",
|
|
(guint) info.bytesconsumed, (guint) info.samples);
|
|
|
|
if (info.bytesconsumed > input_size)
|
|
info.bytesconsumed = input_size;
|
|
|
|
input_size -= info.bytesconsumed;
|
|
input_data += info.bytesconsumed;
|
|
|
|
if (out && info.samples > 0) {
|
|
if (!gst_faad_update_caps (faad, &info))
|
|
goto negotiation_failed;
|
|
|
|
/* C's lovely propensity for int overflow.. */
|
|
if (info.samples > G_MAXUINT / faad->bps)
|
|
goto sample_overflow;
|
|
|
|
/* play decoded data */
|
|
if (info.samples > 0) {
|
|
guint bufsize = info.samples * faad->bps;
|
|
guint num_samples = info.samples / faad->channels;
|
|
|
|
/* note: info.samples is total samples, not per channel */
|
|
ret =
|
|
gst_pad_alloc_buffer_and_set_caps (faad->srcpad, 0, bufsize,
|
|
GST_PAD_CAPS (faad->srcpad), &outbuf);
|
|
if (ret != GST_FLOW_OK)
|
|
goto out;
|
|
|
|
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
|
|
GST_BUFFER_OFFSET (outbuf) =
|
|
GST_CLOCK_TIME_TO_FRAMES (faad->next_ts, faad->samplerate);
|
|
GST_BUFFER_TIMESTAMP (outbuf) = faad->next_ts;
|
|
GST_BUFFER_DURATION (outbuf) =
|
|
GST_FRAMES_TO_CLOCK_TIME (num_samples, faad->samplerate);
|
|
|
|
GST_OBJECT_LOCK (faad);
|
|
faad->next_ts += GST_BUFFER_DURATION (outbuf);
|
|
faad->sum_dur_out += GST_BUFFER_DURATION (outbuf);
|
|
faad->bytes_in += info.bytesconsumed;
|
|
GST_OBJECT_UNLOCK (faad);
|
|
|
|
if ((outbuf = gst_audio_buffer_clip (outbuf, &faad->segment,
|
|
faad->samplerate, faad->bps * faad->channels))) {
|
|
GST_LOG_OBJECT (faad,
|
|
"pushing buffer, off=%" G_GUINT64_FORMAT ", ts=%" GST_TIME_FORMAT,
|
|
GST_BUFFER_OFFSET (outbuf),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
|
|
|
|
if (faad->discont) {
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
faad->discont = FALSE;
|
|
}
|
|
|
|
if (faad->segment.rate > 0.0) {
|
|
ret = gst_pad_push (faad->srcpad, outbuf);
|
|
} else {
|
|
/* reverse playback, queue frame till later when we get a discont. */
|
|
GST_LOG_OBJECT (faad, "queued frame");
|
|
faad->queued = g_list_prepend (faad->queued, outbuf);
|
|
ret = GST_FLOW_OK;
|
|
}
|
|
if (ret != GST_FLOW_OK)
|
|
goto out;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
out:
|
|
/* in raw case: (pretend) all consumed */
|
|
if (faad->packetised)
|
|
input_size = 0;
|
|
gst_adapter_flush (faad->adapter, available - input_size);
|
|
|
|
gst_object_unref (faad);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
init_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
|
|
("Failed to init decoder from stream"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
init2_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
|
|
("%s() failed", (faad->handle) ? "faacDecInit2" : "faacDecOpen"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
decode_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
|
|
("decoding error: %s", faacDecGetErrorMessage (info.error)));
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
negotiation_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL),
|
|
("Setting caps on source pad failed"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
sample_overflow:
|
|
{
|
|
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
|
|
("Output buffer too large"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
parse_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
|
|
("failed to parse non-packetized stream"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_faad_open_decoder (GstFaad * faad)
|
|
{
|
|
faacDecConfiguration *conf;
|
|
|
|
faad->handle = faacDecOpen ();
|
|
|
|
if (faad->handle == NULL) {
|
|
GST_WARNING_OBJECT (faad, "faacDecOpen() failed");
|
|
return FALSE;
|
|
}
|
|
|
|
conf = faacDecGetCurrentConfiguration (faad->handle);
|
|
conf->defObjectType = LC;
|
|
conf->dontUpSampleImplicitSBR = 1;
|
|
conf->outputFormat = FAAD_FMT_16BIT;
|
|
|
|
if (faacDecSetConfiguration (faad->handle, conf) == 0) {
|
|
GST_WARNING_OBJECT (faad, "faacDecSetConfiguration() failed");
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_faad_close_decoder (GstFaad * faad)
|
|
{
|
|
if (faad->handle) {
|
|
faacDecClose (faad->handle);
|
|
faad->handle = NULL;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_faad_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstFaad *faad = GST_FAAD (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (!gst_faad_open_decoder (faad))
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_faad_reset (faad);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_faad_close_decoder (faad);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"faad",
|
|
"Free AAC Decoder (FAAD)",
|
|
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|