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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1b2fa58301
Original commit message from CVS: * ext/faac/gstfaac.c: (gst_faac_sink_event): Don't try to flush the decoder on EOS when it was not initialized. Fixes #498667
768 lines
22 KiB
C
768 lines
22 KiB
C
/* 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 <string.h>
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#include "gstfaac.h"
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#define SINK_CAPS \
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"audio/x-raw-int, " \
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"endianness = (int) BYTE_ORDER, " \
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"signed = (boolean) true, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"rate = (int) [ 8000, 96000 ], " \
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"channels = (int) [ 1, 6 ] "
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/* these don't seem to work? */
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#if 0
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"audio/x-raw-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, "
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"width = (int) 32, "
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"depth = (int) { 24, 32 }, "
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"rate = (int) [ 8000, 96000], "
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"channels = (int) [ 1, 6]; "
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"audio/x-raw-float, "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, "
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"rate = (int) [ 8000, 96000], " "channels = (int) [ 1, 6]"
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#endif
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#define SRC_CAPS \
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"audio/mpeg, " \
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"mpegversion = (int) { 4, 2 }, " \
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"channels = (int) [ 1, 6 ], " \
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"rate = (int) [ 8000, 96000 ]"
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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 (SRC_CAPS));
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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 (SINK_CAPS));
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static const GstElementDetails gst_faac_details =
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GST_ELEMENT_DETAILS ("AAC audio encoder",
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"Codec/Encoder/Audio",
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"Free MPEG-2/4 AAC encoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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enum
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{
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ARG_0,
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ARG_OUTPUTFORMAT,
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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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};
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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, const GValue * value, GParamSpec * pspec);
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static void gst_faac_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_faac_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_faac_chain (GstPad * pad, GstBuffer * data);
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static GstStateChangeReturn gst_faac_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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GST_DEBUG_CATEGORY_STATIC (faac_debug);
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#define GST_CAT_DEFAULT faac_debug
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#define FAAC_DEFAULT_MPEGVERSION 4
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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", &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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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 (element_class, &gst_faac_details);
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GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
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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, "LC", "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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#define GST_TYPE_FAAC_OUTPUTFORMAT (gst_faac_outputformat_get_type ())
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static GType
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gst_faac_outputformat_get_type (void)
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{
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static GType gst_faac_outputformat_type = 0;
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if (!gst_faac_outputformat_type) {
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static GEnumValue gst_faac_outputformat[] = {
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{0, "OUTPUTFORMAT_RAW", "Raw AAC"},
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{1, "OUTPUTFORMAT_ADTS", "ADTS headers"},
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{0, NULL, NULL},
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};
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gst_faac_outputformat_type = g_enum_register_static ("GstFaacOutputFormat",
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gst_faac_outputformat);
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}
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return gst_faac_outputformat_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_peek_parent (klass);
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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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/* 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 * 1000, 320 * 1000, 128 * 1000, 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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g_object_class_install_property (gobject_class, ARG_OUTPUTFORMAT,
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g_param_spec_enum ("outputformat", "Output format",
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"Format of output frames",
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GST_TYPE_FAAC_OUTPUTFORMAT, MAIN, G_PARAM_READWRITE));
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/* virtual functions */
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faac_change_state);
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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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faac->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_chain_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_chain));
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gst_pad_set_setcaps_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_sink_setcaps));
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gst_pad_set_event_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_sink_event));
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gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
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faac->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (faac->srcpad);
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gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
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/* default properties */
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faac->bitrate = 1000 * 128;
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faac->profile = MAIN;
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faac->shortctl = SHORTCTL_NORMAL;
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faac->outputformat = 0;
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faac->tns = FALSE;
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faac->midside = TRUE;
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}
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static gboolean
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gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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faacEncHandle *handle;
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gint channels, samplerate, width;
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gulong samples, bytes, fmt = 0, bps = 0;
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gboolean result = FALSE;
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if (!gst_caps_is_fixed (caps))
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goto done;
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GST_OBJECT_LOCK (faac);
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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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GST_OBJECT_UNLOCK (faac);
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if (!gst_structure_get_int (structure, "channels", &channels) ||
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!gst_structure_get_int (structure, "rate", &samplerate)) {
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goto done;
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}
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if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes)))
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goto done;
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if (gst_structure_has_name (structure, "audio/x-raw-int")) {
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gst_structure_get_int (structure, "width", &width);
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switch (width) {
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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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case 32:
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fmt = FAAC_INPUT_32BIT;
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bps = 4;
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break;
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default:
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g_return_val_if_reached (FALSE);
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}
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} else if (gst_structure_has_name (structure, "audio/x-raw-float")) {
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fmt = FAAC_INPUT_FLOAT;
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bps = 4;
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}
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if (!fmt) {
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faacEncClose (handle);
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goto done;
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}
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GST_OBJECT_LOCK (faac);
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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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GST_OBJECT_UNLOCK (faac);
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result = TRUE;
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done:
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gst_object_unref (faac);
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return result;
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}
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static gboolean
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gst_faac_configure_source_pad (GstFaac * faac)
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{
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GstCaps *allowed_caps;
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GstCaps *srccaps;
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gboolean ret = FALSE;
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gint n, ver, mpegversion;
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faacEncConfiguration *conf;
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mpegversion = FAAC_DEFAULT_MPEGVERSION;
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allowed_caps = gst_pad_get_allowed_caps (faac->srcpad);
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GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
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if (allowed_caps == NULL)
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return FALSE;
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if (gst_caps_is_empty (allowed_caps))
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goto empty_caps;
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if (!gst_caps_is_any (allowed_caps)) {
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for (n = 0; n < gst_caps_get_size (allowed_caps); n++) {
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GstStructure *s = gst_caps_get_structure (allowed_caps, n);
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if (gst_structure_get_int (s, "mpegversion", &ver) &&
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(ver == 4 || ver == 2)) {
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mpegversion = ver;
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break;
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}
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}
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}
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gst_caps_unref (allowed_caps);
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/* we negotiated caps update current configuration */
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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 / faac->channels;
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conf->inputFormat = faac->format;
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conf->outputFormat = faac->outputformat;
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conf->shortctl = faac->shortctl;
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if (!faacEncSetConfiguration (faac->handle, conf))
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goto set_failed;
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/* now create a caps for it all */
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srccaps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, mpegversion,
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"channels", G_TYPE_INT, faac->channels,
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"rate", G_TYPE_INT, faac->samplerate, NULL);
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if (mpegversion == 4) {
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GstBuffer *codec_data;
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guint8 *config = NULL;
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gulong config_len = 0;
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/* get the config string */
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GST_DEBUG_OBJECT (faac, "retrieving decoder info");
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faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len);
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/* copy it into a buffer */
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codec_data = gst_buffer_new_and_alloc (config_len);
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memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
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/* add to caps */
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gst_caps_set_simple (srccaps,
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"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
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gst_buffer_unref (codec_data);
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}
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GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps);
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ret = gst_pad_set_caps (faac->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return ret;
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/* ERROR */
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empty_caps:
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{
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gst_caps_unref (allowed_caps);
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return FALSE;
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}
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set_failed:
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{
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GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration");
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return FALSE;
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}
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}
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static gboolean
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gst_faac_sink_event (GstPad * pad, GstEvent * event)
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{
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GstFaac *faac;
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gboolean ret;
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faac = GST_FAAC (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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{
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GstBuffer *outbuf;
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if (!faac->handle)
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ret = FALSE;
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else
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ret = TRUE;
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/* flush first */
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while (ret) {
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if (gst_pad_alloc_buffer_and_set_caps (faac->srcpad,
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GST_BUFFER_OFFSET_NONE, faac->bytes,
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GST_PAD_CAPS (faac->srcpad), &outbuf) == GST_FLOW_OK) {
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gint ret_size;
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if ((ret_size = faacEncEncode (faac->handle, NULL, 0,
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GST_BUFFER_DATA (outbuf), faac->bytes)) > 0) {
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GST_BUFFER_SIZE (outbuf) = ret_size;
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GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
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GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
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gst_pad_push (faac->srcpad, outbuf);
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|
} else {
|
|
gst_buffer_unref (outbuf);
|
|
ret = FALSE;
|
|
}
|
|
}
|
|
}
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_NEWSEGMENT:
|
|
ret = gst_pad_push_event (faac->srcpad, event);
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
default:
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
|
|
}
|
|
gst_object_unref (faac);
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_faac_chain (GstPad * pad, GstBuffer * inbuf)
|
|
{
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
GstBuffer *outbuf, *subbuf;
|
|
GstFaac *faac;
|
|
guint size, ret_size, in_size, frame_size;
|
|
|
|
faac = GST_FAAC (gst_pad_get_parent (pad));
|
|
|
|
if (!faac->handle)
|
|
goto no_handle;
|
|
|
|
if (!GST_PAD_CAPS (faac->srcpad)) {
|
|
if (!gst_faac_configure_source_pad (faac))
|
|
goto nego_failed;
|
|
}
|
|
|
|
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) {
|
|
GstBuffer *merge;
|
|
|
|
/* this is panic! we got a buffer, but still don't have enough
|
|
* data. Merge them and retry in the next cycle... */
|
|
merge = gst_buffer_merge (faac->cache, inbuf);
|
|
gst_buffer_unref (faac->cache);
|
|
gst_buffer_unref (inbuf);
|
|
faac->cache = merge;
|
|
} 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);
|
|
}
|
|
|
|
goto done;
|
|
}
|
|
|
|
/* create the frame */
|
|
if (in_size > size) {
|
|
GstBuffer *merge;
|
|
|
|
/* 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;
|
|
merge = gst_buffer_merge (faac->cache, subbuf);
|
|
gst_buffer_unref (faac->cache);
|
|
gst_buffer_unref (subbuf);
|
|
subbuf = merge;
|
|
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);
|
|
}
|
|
|
|
result =
|
|
gst_pad_alloc_buffer_and_set_caps (faac->srcpad, GST_BUFFER_OFFSET_NONE,
|
|
faac->bytes, GST_PAD_CAPS (faac->srcpad), &outbuf);
|
|
if (result != GST_FLOW_OK)
|
|
goto done;
|
|
|
|
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 (faac, LIBRARY, ENCODE, (NULL), (NULL));
|
|
gst_buffer_unref (inbuf);
|
|
gst_buffer_unref (subbuf);
|
|
result = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
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;
|
|
}
|
|
result = gst_pad_push (faac->srcpad, 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);
|
|
}
|
|
|
|
done:
|
|
gst_object_unref (faac);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_handle:
|
|
{
|
|
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
|
|
("format wasn't negotiated before chain function"));
|
|
gst_buffer_unref (inbuf);
|
|
result = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
nego_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
|
|
("failed to negotiate MPEG/AAC format with next element"));
|
|
gst_buffer_unref (inbuf);
|
|
result = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_faac_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstFaac *faac = GST_FAAC (object);
|
|
|
|
GST_OBJECT_LOCK (faac);
|
|
|
|
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;
|
|
case ARG_OUTPUTFORMAT:
|
|
faac->outputformat = g_value_get_enum (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (faac);
|
|
}
|
|
|
|
static void
|
|
gst_faac_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstFaac *faac = GST_FAAC (object);
|
|
|
|
GST_OBJECT_LOCK (faac);
|
|
|
|
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;
|
|
case ARG_OUTPUTFORMAT:
|
|
g_value_set_enum (value, faac->outputformat);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (faac);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_faac_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstFaac *faac = GST_FAAC (element);
|
|
|
|
/* upwards state changes */
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
/* downwards state changes */
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
{
|
|
GST_OBJECT_LOCK (faac);
|
|
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;
|
|
GST_OBJECT_UNLOCK (faac);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
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_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|