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Actually, FAAC is LGPL, not GPL (like FAAD)
Original commit message from CVS: Actually, FAAC is LGPL, not GPL (like FAAD)
This commit is contained in:
Ronald S. Bultje
parent
8d529948e6
commit
839f6e7525
1 changed files with 680 additions and 0 deletions
680
ext/faac/gstfaac.c
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680
ext/faac/gstfaac.c
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/* GStreamer FAAC (Free AAC Encoder) plugin
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* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.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 "gstfaac.h"
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GST_PAD_TEMPLATE_FACTORY (src_template,
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"src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"faac_mpeg_templ",
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"audio/mpeg",
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"systemstream", GST_PROPS_BOOLEAN (FALSE),
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"mpegversion", GST_PROPS_LIST (
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GST_PROPS_INT (4), /* we prefer 4 */
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GST_PROPS_INT (2)
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),
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"channels", GST_PROPS_INT_RANGE (1, 6),
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"samplerate", GST_PROPS_INT_RANGE (8000, 96000)
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)
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);
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GST_PAD_TEMPLATE_FACTORY (sink_template,
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"faac_int16_templ",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_BOOLEAN (TRUE),
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"width", GST_PROPS_INT (16),
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"depth", GST_PROPS_INT (16),
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"rate", GST_PROPS_INT_RANGE (8000, 96000),
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"channels", GST_PROPS_INT_RANGE (1, 6)
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),
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GST_CAPS_NEW (
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"faac_int24_templ",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_BOOLEAN (TRUE),
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"width", GST_PROPS_INT (32),
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"depth", GST_PROPS_INT (24),
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"rate", GST_PROPS_INT_RANGE (8000, 96000),
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"channels", GST_PROPS_INT_RANGE (1, 6)
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),
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GST_CAPS_NEW (
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"faac_float_templ",
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"audio/x-raw-float",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"depth", GST_PROPS_INT (32), /* float */
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"rate", GST_PROPS_INT_RANGE (8000, 96000),
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"channels", GST_PROPS_INT_RANGE (1, 6)
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)
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);
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enum {
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ARG_0,
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ARG_BITRATE,
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ARG_PROFILE,
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ARG_TNS,
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ARG_MIDSIDE,
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ARG_SHORTCTL
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/* FILL ME */
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};
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static void gst_faac_base_init (GstFaacClass *klass);
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static void gst_faac_class_init (GstFaacClass *klass);
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static void gst_faac_init (GstFaac *faac);
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static void gst_faac_set_property (GObject *object,
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guint prop_id,
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const GValue *value,
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GParamSpec *pspec);
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static void gst_faac_get_property (GObject *object,
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guint prop_id,
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GValue *value,
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GParamSpec *pspec);
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static GstPadLinkReturn
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gst_faac_sinkconnect (GstPad *pad,
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GstCaps *caps);
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static GstPadLinkReturn
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gst_faac_srcconnect (GstPad *pad,
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GstCaps *caps);
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static void gst_faac_chain (GstPad *pad,
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GstData *data);
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static GstElementStateReturn
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gst_faac_change_state (GstElement *element);
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static GstElementClass *parent_class = NULL;
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/* static guint gst_faac_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_faac_get_type (void)
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{
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static GType gst_faac_type = 0;
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if (!gst_faac_type) {
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static const GTypeInfo gst_faac_info = {
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sizeof (GstFaacClass),
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(GBaseInitFunc) gst_faac_base_init,
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NULL,
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(GClassInitFunc) gst_faac_class_init,
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NULL,
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NULL,
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sizeof(GstFaac),
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0,
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(GInstanceInitFunc) gst_faac_init,
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};
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gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstFaac",
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&gst_faac_info, 0);
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}
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return gst_faac_type;
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}
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static void
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gst_faac_base_init (GstFaacClass *klass)
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{
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GstElementDetails gst_faac_details = {
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"Free AAC Encoder (FAAC)",
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"Codec/Audio/Encoder",
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"Free MPEG-2/4 AAC encoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>",
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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_PAD_TEMPLATE_GET (src_template));
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gst_element_class_add_pad_template (element_class,
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GST_PAD_TEMPLATE_GET (sink_template));
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gst_element_class_set_details (element_class, &gst_faac_details);
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}
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#define GST_TYPE_FAAC_PROFILE (gst_faac_profile_get_type ())
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static GType
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gst_faac_profile_get_type (void)
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{
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static GType gst_faac_profile_type = 0;
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if (!gst_faac_profile_type) {
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static GEnumValue gst_faac_profile[] = {
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{ MAIN, "MAIN", "Main profile" },
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{ LOW, "LOW", "Low complexity profile" },
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{ SSR, "SSR", "Scalable sampling rate profile" },
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{ LTP, "LTP", "Long term prediction profile" },
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{ 0, NULL, NULL },
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};
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gst_faac_profile_type = g_enum_register_static ("GstFaacProfile",
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gst_faac_profile);
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}
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return gst_faac_profile_type;
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}
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#define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ())
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static GType
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gst_faac_shortctl_get_type (void)
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{
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static GType gst_faac_shortctl_type = 0;
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if (!gst_faac_shortctl_type) {
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static GEnumValue gst_faac_shortctl[] = {
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{ SHORTCTL_NORMAL, "SHORTCTL_NORMAL", "Normal block type" },
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{ SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks" },
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{ SHORTCTL_NOLONG, "SHORTCTL_NOLONG", "No long blocks" },
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{ 0, NULL, NULL },
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};
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gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl",
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gst_faac_shortctl);
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}
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return gst_faac_shortctl_type;
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}
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static void
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gst_faac_class_init (GstFaacClass *klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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/* properties */
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g_object_class_install_property (gobject_class, ARG_BITRATE,
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g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
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8 * 1024, 320 * 1024, 128 * 1024, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_PROFILE,
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g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile",
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GST_TYPE_FAAC_PROFILE, MAIN, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_TNS,
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g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
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FALSE, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_MIDSIDE,
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g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
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TRUE, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_SHORTCTL,
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g_param_spec_enum ("shortctl", "Block type",
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"Block type encorcing",
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GST_TYPE_FAAC_SHORTCTL, MAIN, G_PARAM_READWRITE));
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/* virtual functions */
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gstelement_class->change_state = gst_faac_change_state;
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gobject_class->set_property = gst_faac_set_property;
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gobject_class->get_property = gst_faac_get_property;
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}
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static void
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gst_faac_init (GstFaac *faac)
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{
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faac->handle = NULL;
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faac->samplerate = -1;
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faac->channels = -1;
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faac->cache = NULL;
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faac->cache_time = GST_CLOCK_TIME_NONE;
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faac->cache_duration = 0;
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GST_FLAG_SET (faac, GST_ELEMENT_EVENT_AWARE);
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faac->sinkpad = gst_pad_new_from_template (
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GST_PAD_TEMPLATE_GET (sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
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gst_pad_set_chain_function (faac->sinkpad, gst_faac_chain);
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gst_pad_set_link_function (faac->sinkpad, gst_faac_sinkconnect);
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faac->srcpad = gst_pad_new_from_template (
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GST_PAD_TEMPLATE_GET (src_template), "src");
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gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
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gst_pad_set_link_function (faac->srcpad, gst_faac_srcconnect);
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/* default properties */
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faac->bitrate = 1024 * 128;
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faac->profile = MAIN;
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faac->shortctl = SHORTCTL_NORMAL;
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faac->tns = FALSE;
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faac->midside = TRUE;
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}
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static GstPadLinkReturn
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gst_faac_sinkconnect (GstPad *pad,
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GstCaps *caps)
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{
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GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
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if (!GST_CAPS_IS_FIXED (caps))
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return GST_PAD_LINK_DELAYED;
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if (faac->handle) {
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faacEncClose (faac->handle);
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faac->handle = NULL;
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}
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if (faac->cache) {
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gst_buffer_unref (faac->cache);
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faac->cache = NULL;
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}
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for (; caps != NULL; caps = caps->next) {
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faacEncHandle *handle;
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gint channels, samplerate, depth;
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gulong samples, bytes, fmt = 0, bps = 0;
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gst_caps_get (caps, "channels", &channels,
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"rate", &samplerate,
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"depth", &depth, NULL);
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/* open a new handle to the encoder */
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if (!(handle = faacEncOpen (samplerate, channels,
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&samples, &bytes)))
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continue;
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switch (depth) {
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case 16:
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fmt = FAAC_INPUT_16BIT;
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bps = 2;
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break;
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case 24:
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fmt = FAAC_INPUT_32BIT; /* 24-in-32, actually */
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bps = 4;
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break;
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case 32:
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fmt = FAAC_INPUT_FLOAT; /* see template, this is right */
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bps = 4;
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break;
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}
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if (!fmt) {
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faacEncClose (handle);
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continue;
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}
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faac->format = fmt;
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faac->bps = bps;
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faac->handle = handle;
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faac->bytes = bytes;
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faac->samples = samples;
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faac->channels = channels;
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faac->samplerate = samplerate;
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/* if the other side was already set-up, redo that */
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if (GST_PAD_CAPS (faac->srcpad))
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return gst_faac_srcconnect (faac->srcpad,
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gst_pad_get_allowed_caps (faac->srcpad));
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/* else, that'll be done later */
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return GST_PAD_LINK_OK;
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}
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return GST_PAD_LINK_REFUSED;
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}
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static GstPadLinkReturn
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gst_faac_srcconnect (GstPad *pad,
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GstCaps *caps)
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{
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GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
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GstCaps *t;
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if (!faac->handle ||
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(faac->samplerate == -1 || faac->channels == -1)) {
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return GST_PAD_LINK_DELAYED;
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}
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/* we do samplerate/channels ourselves */
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for (t = caps; t != NULL; t = t->next) {
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gst_props_remove_entry_by_name (t->properties, "rate");
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gst_props_remove_entry_by_name (t->properties, "channels");
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}
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/* go through list */
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caps = gst_caps_normalize (caps);
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for ( ; caps != NULL; caps = caps->next) {
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faacEncConfiguration *conf;
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gint mpegversion = 0;
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GstCaps *newcaps;
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GstPadLinkReturn ret;
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gst_caps_get_int (caps, "mpegversion", &mpegversion);
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/* new conf */
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conf = faacEncGetCurrentConfiguration (faac->handle);
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conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
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conf->aacObjectType = faac->profile;
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conf->allowMidside = faac->midside;
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conf->useLfe = 0;
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conf->useTns = faac->tns;
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conf->bitRate = faac->bitrate;
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conf->inputFormat = faac->format;
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/* FIXME: this one here means that we do not support direct
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* "MPEG audio file" output (like mp3). This means we can
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* only mux this into mov/qt (mp4a) or matroska or so. If
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* we want to support direct AAC file output, we need ADTS
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* headers, and we need to find a way in the caps to detect
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* that (that the next element is filesink or any element
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* that does want ADTS headers). */
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conf->outputFormat = 0; /* raw, no ADTS headers */
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conf->shortctl = faac->shortctl;
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if (!faacEncSetConfiguration (faac->handle, conf)) {
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GST_WARNING ("Faac doesn't support the current conf");
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continue;
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}
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newcaps = GST_CAPS_NEW ("faac_mpeg_caps",
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"audio/mpeg",
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"systemstream", GST_PROPS_BOOLEAN (FALSE),
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"mpegversion", GST_PROPS_INT (mpegversion),
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"channels", GST_PROPS_INT (faac->channels),
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"rate", GST_PROPS_INT (faac->samplerate));
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ret = gst_pad_try_set_caps (faac->srcpad, newcaps);
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switch (ret) {
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case GST_PAD_LINK_OK:
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case GST_PAD_LINK_DONE:
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return GST_PAD_LINK_DONE;
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case GST_PAD_LINK_DELAYED:
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return GST_PAD_LINK_DELAYED;
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default:
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break;
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}
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}
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return GST_PAD_LINK_REFUSED;
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}
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static void
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gst_faac_chain (GstPad *pad,
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GstData *data)
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{
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GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
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GstBuffer *inbuf, *outbuf, *subbuf;
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guint size, ret_size, in_size, frame_size;
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if (GST_IS_EVENT (data)) {
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GstEvent *event = GST_EVENT (data);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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/* flush first */
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while (1) {
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outbuf = gst_buffer_new_and_alloc (faac->bytes);
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if ((ret_size = faacEncEncode (faac->handle,
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NULL, 0,
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GST_BUFFER_DATA (outbuf),
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faac->bytes)) < 0) {
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gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
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gst_event_unref (event);
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gst_buffer_unref (outbuf);
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return;
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}
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if (ret_size > 0) {
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GST_BUFFER_SIZE (outbuf) = ret_size;
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GST_BUFFER_TIMESTAMP (outbuf) = 0;
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GST_BUFFER_DURATION (outbuf) = 0;
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gst_pad_push (faac->srcpad, GST_DATA (outbuf));
|
||||
} else {
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
gst_element_set_eos (GST_ELEMENT (faac));
|
||||
gst_pad_push (faac->srcpad, data);
|
||||
return;
|
||||
default:
|
||||
gst_pad_event_default (pad, event);
|
||||
return;
|
||||
}
|
||||
}
|
||||
|
||||
inbuf = GST_BUFFER (data);
|
||||
|
||||
if (!faac->handle) {
|
||||
gst_element_error (GST_ELEMENT (faac),
|
||||
"No input format negotiated");
|
||||
gst_buffer_unref (inbuf);
|
||||
return;
|
||||
}
|
||||
|
||||
if (!GST_PAD_CAPS (faac->srcpad)) {
|
||||
if (gst_faac_srcconnect (faac->srcpad,
|
||||
gst_pad_get_allowed_caps (faac->srcpad)) <= 0) {
|
||||
gst_element_error (GST_ELEMENT (faac),
|
||||
"Failed to negotiate MPEG/AAC format with next element");
|
||||
gst_buffer_unref (inbuf);
|
||||
return;
|
||||
}
|
||||
}
|
||||
|
||||
size = GST_BUFFER_SIZE (inbuf);
|
||||
in_size = size;
|
||||
if (faac->cache)
|
||||
in_size += GST_BUFFER_SIZE (faac->cache);
|
||||
frame_size = faac->samples * faac->bps;
|
||||
|
||||
while (1) {
|
||||
/* do we have enough data for one frame? */
|
||||
if (in_size / faac->bps < faac->samples) {
|
||||
if (in_size > size) {
|
||||
/* this is panic! we got a buffer, but still don't have enough
|
||||
* data. Merge them and retry in the next cycle... */
|
||||
faac->cache = gst_buffer_merge (faac->cache, inbuf);
|
||||
} else if (in_size == size) {
|
||||
/* this shouldn't happen, but still... */
|
||||
faac->cache = inbuf;
|
||||
} else if (in_size > 0) {
|
||||
faac->cache = gst_buffer_create_sub (inbuf, size - in_size,
|
||||
in_size);
|
||||
GST_BUFFER_DURATION (faac->cache) =
|
||||
GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (faac->cache) / size;
|
||||
GST_BUFFER_TIMESTAMP (faac->cache) =
|
||||
GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
|
||||
(size - in_size) / size);
|
||||
gst_buffer_unref (inbuf);
|
||||
} else {
|
||||
gst_buffer_unref (inbuf);
|
||||
}
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
/* create the frame */
|
||||
if (in_size > size) {
|
||||
/* merge */
|
||||
subbuf = gst_buffer_create_sub (inbuf, 0, frame_size - (in_size - size));
|
||||
GST_BUFFER_DURATION (subbuf) =
|
||||
GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
|
||||
subbuf = gst_buffer_merge (faac->cache, subbuf);
|
||||
faac->cache = NULL;
|
||||
} else {
|
||||
subbuf = gst_buffer_create_sub (inbuf, size - in_size, frame_size);
|
||||
GST_BUFFER_DURATION (subbuf) =
|
||||
GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
|
||||
GST_BUFFER_TIMESTAMP (subbuf) =
|
||||
GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
|
||||
(size - in_size) / size);
|
||||
}
|
||||
|
||||
outbuf = gst_buffer_new_and_alloc (faac->bytes);
|
||||
if ((ret_size = faacEncEncode (faac->handle,
|
||||
(gint32 *) GST_BUFFER_DATA (subbuf),
|
||||
GST_BUFFER_SIZE (subbuf) / faac->bps,
|
||||
GST_BUFFER_DATA (outbuf),
|
||||
faac->bytes)) < 0) {
|
||||
gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
|
||||
gst_buffer_unref (inbuf);
|
||||
gst_buffer_unref (subbuf);
|
||||
return;
|
||||
}
|
||||
|
||||
if (ret_size > 0) {
|
||||
GST_BUFFER_SIZE (outbuf) = ret_size;
|
||||
if (faac->cache_time != GST_CLOCK_TIME_NONE) {
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = faac->cache_time;
|
||||
faac->cache_time = GST_CLOCK_TIME_NONE;
|
||||
} else
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (subbuf);
|
||||
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (subbuf);
|
||||
if (faac->cache_duration) {
|
||||
GST_BUFFER_DURATION (outbuf) += faac->cache_duration;
|
||||
faac->cache_duration = 0;
|
||||
}
|
||||
gst_pad_push (faac->srcpad, GST_DATA (outbuf));
|
||||
} else {
|
||||
/* FIXME: what I'm doing here isn't fully correct, but there
|
||||
* really isn't a better way yet.
|
||||
* Problem is that libfaac caches buffers (for encoding
|
||||
* purposes), so the timestamp of the outgoing buffer isn't
|
||||
* the same as the timestamp of the data that I pushed in.
|
||||
* However, I don't know the delay between those two so I
|
||||
* cannot really say aything about it. This is a bad guess. */
|
||||
|
||||
gst_buffer_unref (outbuf);
|
||||
if (faac->cache_time != GST_CLOCK_TIME_NONE)
|
||||
faac->cache_time = GST_BUFFER_TIMESTAMP (subbuf);
|
||||
faac->cache_duration += GST_BUFFER_DURATION (subbuf);
|
||||
}
|
||||
|
||||
in_size -= frame_size;
|
||||
gst_buffer_unref (subbuf);
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
gst_faac_set_property (GObject *object,
|
||||
guint prop_id,
|
||||
const GValue *value,
|
||||
GParamSpec *pspec)
|
||||
{
|
||||
GstFaac *faac = GST_FAAC (object);
|
||||
|
||||
switch (prop_id) {
|
||||
case ARG_BITRATE:
|
||||
faac->bitrate = g_value_get_int (value);
|
||||
break;
|
||||
case ARG_PROFILE:
|
||||
faac->profile = g_value_get_enum (value);
|
||||
break;
|
||||
case ARG_TNS:
|
||||
faac->tns = g_value_get_boolean (value);
|
||||
break;
|
||||
case ARG_MIDSIDE:
|
||||
faac->midside = g_value_get_boolean (value);
|
||||
break;
|
||||
case ARG_SHORTCTL:
|
||||
faac->shortctl = g_value_get_enum (value);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
gst_faac_get_property (GObject *object,
|
||||
guint prop_id,
|
||||
GValue *value,
|
||||
GParamSpec *pspec)
|
||||
{
|
||||
GstFaac *faac = GST_FAAC (object);
|
||||
|
||||
switch (prop_id) {
|
||||
case ARG_BITRATE:
|
||||
g_value_set_int (value, faac->bitrate);
|
||||
break;
|
||||
case ARG_PROFILE:
|
||||
g_value_set_enum (value, faac->profile);
|
||||
break;
|
||||
case ARG_TNS:
|
||||
g_value_set_boolean (value, faac->tns);
|
||||
break;
|
||||
case ARG_MIDSIDE:
|
||||
g_value_set_boolean (value, faac->midside);
|
||||
break;
|
||||
case ARG_SHORTCTL:
|
||||
g_value_set_enum (value, faac->shortctl);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
static GstElementStateReturn
|
||||
gst_faac_change_state (GstElement *element)
|
||||
{
|
||||
GstFaac *faac = GST_FAAC (element);
|
||||
|
||||
switch (GST_STATE_TRANSITION (element)) {
|
||||
case GST_STATE_PAUSED_TO_READY:
|
||||
if (faac->handle) {
|
||||
faacEncClose (faac->handle);
|
||||
faac->handle = NULL;
|
||||
}
|
||||
if (faac->cache) {
|
||||
gst_buffer_unref (faac->cache);
|
||||
faac->cache = NULL;
|
||||
}
|
||||
faac->cache_time = GST_CLOCK_TIME_NONE;
|
||||
faac->cache_duration = 0;
|
||||
faac->samplerate = -1;
|
||||
faac->channels = -1;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
||||
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
||||
|
||||
return GST_STATE_SUCCESS;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
plugin_init (GstPlugin *plugin)
|
||||
{
|
||||
return gst_element_register (plugin, "faac",
|
||||
GST_RANK_NONE,
|
||||
GST_TYPE_FAAC);
|
||||
}
|
||||
|
||||
GST_PLUGIN_DEFINE (
|
||||
GST_VERSION_MAJOR,
|
||||
GST_VERSION_MINOR,
|
||||
"faac",
|
||||
"Free AAC Encoder (FAAC)",
|
||||
plugin_init,
|
||||
VERSION,
|
||||
"LGPL",
|
||||
GST_COPYRIGHT,
|
||||
GST_PACKAGE,
|
||||
GST_ORIGIN
|
||||
)
|
||||
Loading…
Reference in a new issue