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bf3fd4f95d
Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/arts/gst_arts.c: * ext/artsd/gstartsdsink.c: * ext/audiofile/gstafparse.c: * ext/audiofile/gstafsink.c: * ext/audiofile/gstafsrc.c: * ext/audioresample/gstaudioresample.c: * ext/bz2/gstbz2dec.c: * ext/bz2/gstbz2enc.c: * ext/cdaudio/gstcdaudio.c: * ext/directfb/dfbvideosink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/libfame/gstlibfame.c: * ext/libmms/gstmms.c: (gst_mms_base_init): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/nas/nassink.c: (gst_nassink_base_init): * ext/neon/gstneonhttpsrc.c: * ext/sdl/sdlaudiosink.c: * ext/sdl/sdlvideosink.c: * ext/shout/gstshout.c: * ext/snapshot/gstsnapshot.c: * ext/sndfile/gstsf.c: * ext/swfdec/gstswfdec.c: * ext/tarkin/gsttarkindec.c: * ext/tarkin/gsttarkinenc.c: * ext/theora/theoradec.c: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init): * gst/chart/gstchart.c: * gst/colorspace/gstcolorspace.c: * gst/deinterlace/gstdeinterlace.c: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init): * gst/festival/gstfestival.c: * gst/filter/gstbpwsinc.c: * gst/filter/gstiir.c: * gst/filter/gstlpwsinc.c: * gst/freeze/gstfreeze.c: * gst/games/gstpuzzle.c: (gst_puzzle_base_init): * gst/librfb/gstrfbsrc.c: * gst/mixmatrix/mixmatrix.c: * gst/mpeg1sys/gstmpeg1systemencode.c: * gst/mpeg1videoparse/gstmp1videoparse.c: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/multifilesink/gstmultifilesink.c: * gst/overlay/gstoverlay.c: * gst/passthrough/gstpassthrough.c: * gst/playondemand/gstplayondemand.c: * gst/qtdemux/qtdemux.c: * gst/rtjpeg/gstrtjpegdec.c: * gst/rtjpeg/gstrtjpegenc.c: * gst/smooth/gstsmooth.c: * gst/smoothwave/gstsmoothwave.c: * gst/spectrum/gstspectrum.c: * gst/speed/gstspeed.c: * gst/stereo/gststereo.c: * gst/switch/gstswitch.c: * gst/tta/gstttadec.c: (gst_tta_dec_base_init): * gst/tta/gstttaparse.c: (gst_tta_parse_base_init): * gst/vbidec/gstvbidec.c: * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/virtualdub/gstxsharpen.c: * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init): * gst/y4m/gsty4mencode.c: * sys/cdrom/gstcdplayer.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: * sys/glsink/glimagesink.c: * sys/qcam/gstqcamsrc.c: * sys/v4l2/gstv4l2src.c: * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init): * sys/ximagesrc/ximagesrc.c: Define GstElementDetails as const and also static (when defined as global)
734 lines
21 KiB
C
734 lines
21 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 "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, "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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#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 * 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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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 =
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gst_pad_new_from_template (gst_static_pad_template_get (&sink_template),
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"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 =
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gst_pad_new_from_template (gst_static_pad_template_get (&src_template),
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"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 = 1024 * 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 *src_caps;
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gboolean ret = FALSE;
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gint n, ver, mpegversion;
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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 done;
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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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src_caps = gst_caps_new_simple ("audio/mpeg",
|
|
"mpegversion", G_TYPE_INT, mpegversion,
|
|
"channels", G_TYPE_INT, faac->channels,
|
|
"rate", G_TYPE_INT, faac->samplerate, NULL);
|
|
|
|
GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, src_caps);
|
|
|
|
ret = gst_pad_set_caps (faac->srcpad, src_caps);
|
|
gst_caps_unref (src_caps);
|
|
|
|
if (ret) {
|
|
faacEncConfiguration *conf;
|
|
|
|
/* new conf */
|
|
conf = faacEncGetCurrentConfiguration (faac->handle);
|
|
conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
|
|
conf->aacObjectType = faac->profile;
|
|
conf->allowMidside = faac->midside;
|
|
conf->useLfe = 0;
|
|
conf->useTns = faac->tns;
|
|
conf->bitRate = faac->bitrate;
|
|
conf->inputFormat = faac->format;
|
|
conf->outputFormat = faac->outputformat;
|
|
conf->shortctl = faac->shortctl;
|
|
if (!faacEncSetConfiguration (faac->handle, conf)) {
|
|
GST_WARNING ("Faac doesn't support the current configuration");
|
|
ret = FALSE;
|
|
}
|
|
}
|
|
|
|
done:
|
|
|
|
gst_caps_unref (allowed_caps);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_faac_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstFaac *faac;
|
|
gboolean ret;
|
|
|
|
faac = GST_FAAC (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
{
|
|
GstBuffer *outbuf;
|
|
|
|
/* flush first */
|
|
ret = TRUE;
|
|
do {
|
|
if (gst_pad_alloc_buffer_and_set_caps (faac->srcpad,
|
|
GST_BUFFER_OFFSET_NONE, faac->bytes,
|
|
GST_PAD_CAPS (faac->srcpad), &outbuf) == GST_FLOW_OK) {
|
|
gint ret_size;
|
|
|
|
if ((ret_size = faacEncEncode (faac->handle, NULL, 0,
|
|
GST_BUFFER_DATA (outbuf), faac->bytes)) > 0) {
|
|
GST_BUFFER_SIZE (outbuf) = ret_size;
|
|
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
|
|
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
|
|
gst_pad_push (faac->srcpad, outbuf);
|
|
} else {
|
|
gst_buffer_unref (outbuf);
|
|
ret = FALSE;
|
|
}
|
|
}
|
|
} while (ret);
|
|
|
|
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) {
|
|
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
|
|
("format wasn't negotiated before chain function"));
|
|
gst_buffer_unref (inbuf);
|
|
result = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
if (!GST_PAD_CAPS (faac->srcpad)) {
|
|
if (!gst_faac_configure_source_pad (faac)) {
|
|
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;
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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)
|