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27f2c9b255
Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: (gst_shout2send_init): * ext/shout2/gstshout2.h: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavidemux.c: (gst_avi_demux_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/gstnavseek.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstfdsink.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: Define GstElementDetails as const and also static (when defined as global)
491 lines
13 KiB
C
491 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
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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
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmp4vpay.h"
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GST_DEBUG_CATEGORY (rtpmp4vpay_debug);
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#define GST_CAT_DEFAULT (rtpmp4vpay_debug)
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/* elementfactory information */
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static const GstElementDetails gst_rtp_mp4vpay_details =
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GST_ELEMENT_DETAILS ("RTP packet parser",
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"Codec/Payloader/Network",
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"Payode MPEG4 video as RTP packets (RFC 3016)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_mp4v_pay_sink_template =
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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 ("video/mpeg,"
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"mpegversion=(int) 4," "systemstream=(boolean)false")
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);
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static GstStaticPadTemplate gst_rtp_mp4v_pay_src_template =
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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 ("application/x-rtp, "
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"media = (string) \"video\", "
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"payload = (int) [ 96, 127 ], "
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"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4V-ES\""
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/* two string params
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*
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"profile-level-id = (string) [1,MAX]"
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"config = (string) [1,MAX]"
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*/
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)
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);
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#define DEFAULT_SEND_CONFIG FALSE
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enum
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{
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ARG_0,
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ARG_SEND_CONFIG
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};
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static void gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass);
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static void gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass);
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static void gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay);
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static void gst_rtp_mp4v_pay_finalize (GObject * object);
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static void gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_mp4v_pay_get_type (void)
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{
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static GType rtpmp4vpay_type = 0;
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if (!rtpmp4vpay_type) {
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static const GTypeInfo rtpmp4vpay_info = {
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sizeof (GstRtpMP4VPayClass),
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(GBaseInitFunc) gst_rtp_mp4v_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_mp4v_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpMP4VPay),
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0,
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(GInstanceInitFunc) gst_rtp_mp4v_pay_init,
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};
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rtpmp4vpay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4VPay",
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&rtpmp4vpay_info, 0);
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}
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return rtpmp4vpay_type;
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}
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static void
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gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * 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 (&gst_rtp_mp4v_pay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp4v_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_mp4vpay_details);
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}
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static void
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gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_rtp_mp4v_pay_set_property;
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gobject_class->get_property = gst_rtp_mp4v_pay_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SEND_CONFIG,
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g_param_spec_boolean ("send-config", "Send Config",
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"Send the config parameters in RTP packets as well",
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DEFAULT_SEND_CONFIG, G_PARAM_READWRITE));
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gobject_class->finalize = gst_rtp_mp4v_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_mp4v_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpmp4vpay_debug, "rtpmp4vpay", 0,
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"MP4 video RTP Payloader");
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}
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static void
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gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay)
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{
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rtpmp4vpay->adapter = gst_adapter_new ();
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rtpmp4vpay->rate = 90000;
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rtpmp4vpay->profile = 1;
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rtpmp4vpay->send_config = DEFAULT_SEND_CONFIG;
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}
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static void
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gst_rtp_mp4v_pay_finalize (GObject * object)
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{
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GstRtpMP4VPay *rtpmp4vpay;
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rtpmp4vpay = GST_RTP_MP4V_PAY (object);
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g_object_unref (rtpmp4vpay->adapter);
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rtpmp4vpay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
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{
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gchar *profile, *config;
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GValue v = { 0 };
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profile = g_strdup_printf ("%d", rtpmp4vpay->profile);
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g_value_init (&v, GST_TYPE_BUFFER);
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gst_value_set_buffer (&v, rtpmp4vpay->config);
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config = gst_value_serialize (&v);
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
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"profile-level-id", G_TYPE_STRING, profile,
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"config", G_TYPE_STRING, config, NULL);
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g_value_unset (&v);
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g_free (profile);
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g_free (config);
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}
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static gboolean
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gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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GstRtpMP4VPay *rtpmp4vpay;
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GstStructure *structure;
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const GValue *codec_data;
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rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
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gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
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rtpmp4vpay->rate);
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structure = gst_caps_get_structure (caps, 0);
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codec_data = gst_structure_get_value (structure, "codec_data");
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if (codec_data) {
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GST_LOG_OBJECT (rtpmp4vpay, "got codec_data");
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if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
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GstBuffer *buffer;
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guint8 *data;
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guint size;
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buffer = gst_value_get_buffer (codec_data);
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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if (size < 5)
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goto done;
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rtpmp4vpay->profile = data[4];
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GST_LOG_OBJECT (rtpmp4vpay, "configuring codec_data, profile %d",
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data[4]);
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if (rtpmp4vpay->config)
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gst_buffer_unref (rtpmp4vpay->config);
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rtpmp4vpay->config = gst_buffer_copy (buffer);
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gst_rtp_mp4v_pay_new_caps (rtpmp4vpay);
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}
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}
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done:
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
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{
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guint avail;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. In the case the
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* adapter has more than one MTU, we need to split the MP4V data
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* over multiple packets. */
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avail = gst_adapter_available (rtpmp4vpay->adapter);
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ret = GST_FLOW_OK;
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
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/* fill one MTU or all available bytes */
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towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
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/* this is the payload length */
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payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = (guint8 *) gst_adapter_peek (rtpmp4vpay->adapter, payload_len);
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memcpy (payload, data, payload_len);
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gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
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avail -= payload_len;
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gst_rtp_buffer_set_marker (outbuf, avail == 0);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_ts;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
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}
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return ret;
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}
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#define VOS_STARTCODE 0x000001B0
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#define VOS_ENDCODE 0x000001B1
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#define USER_DATA_STARTCODE 0x000001B2
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#define GOP_STARTCODE 0x000001B3
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#define VISUAL_OBJECT_STARTCODE 0x000001B5
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#define VOP_STARTCODE 0x000001B6
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static gboolean
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gst_rtp_mp4v_pay_depay_data (GstRtpMP4VPay * enc, guint8 * data, guint size,
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gint * strip)
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{
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guint32 code;
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gboolean result;
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*strip = 0;
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if (size < 5)
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return FALSE;
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code = GST_READ_UINT32_BE (data);
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GST_DEBUG_OBJECT (enc, "start code 0x%08x", code);
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switch (code) {
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case VOS_STARTCODE:
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{
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gint i;
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guint8 profile;
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gboolean newprofile = FALSE;
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gboolean equal;
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/* profile_and_level_indication */
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profile = data[4];
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GST_DEBUG_OBJECT (enc, "VOS profile 0x%08x", profile);
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if (profile != enc->profile) {
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newprofile = TRUE;
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enc->profile = profile;
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}
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/* up to the next GOP_STARTCODE or VOP_STARTCODE is
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* the config information */
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code = 0xffffffff;
|
|
for (i = 5; i < size - 4; i++) {
|
|
code = (code << 8) | data[i];
|
|
if (code == GOP_STARTCODE || code == VOP_STARTCODE)
|
|
break;
|
|
}
|
|
i -= 3;
|
|
/* see if config changed */
|
|
equal = FALSE;
|
|
if (enc->config) {
|
|
if (GST_BUFFER_SIZE (enc->config) == i) {
|
|
equal = memcmp (GST_BUFFER_DATA (enc->config), data, i) == 0;
|
|
}
|
|
}
|
|
/* if config string changed or new profile, make new caps */
|
|
if (!equal || newprofile) {
|
|
if (enc->config)
|
|
gst_buffer_unref (enc->config);
|
|
enc->config = gst_buffer_new_and_alloc (i);
|
|
memcpy (GST_BUFFER_DATA (enc->config), data, i);
|
|
gst_rtp_mp4v_pay_new_caps (enc);
|
|
}
|
|
*strip = i;
|
|
/* we need to flush out the current packet. */
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
case VOP_STARTCODE:
|
|
GST_DEBUG_OBJECT (enc, "VOP");
|
|
/* VOP startcode, we don't have to flush the packet */
|
|
result = FALSE;
|
|
break;
|
|
default:
|
|
GST_DEBUG_OBJECT (enc, "other startcode");
|
|
/* all other startcodes need a flush */
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/* we expect buffers starting on startcodes.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpMP4VPay *rtpmp4vpay;
|
|
GstFlowReturn ret;
|
|
guint size, avail;
|
|
guint packet_len;
|
|
guint8 *data;
|
|
gboolean flush;
|
|
gint strip;
|
|
GstClockTime duration;
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload);
|
|
|
|
size = GST_BUFFER_SIZE (buffer);
|
|
data = GST_BUFFER_DATA (buffer);
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
avail = gst_adapter_available (rtpmp4vpay->adapter);
|
|
|
|
/* empty buffer, take timestamp */
|
|
if (avail == 0) {
|
|
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
|
|
rtpmp4vpay->duration = 0;
|
|
}
|
|
|
|
/* depay incomming data and see if we need to start a new RTP
|
|
* packet */
|
|
flush = gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, data, size, &strip);
|
|
if (strip) {
|
|
/* strip off config if requested */
|
|
if (!rtpmp4vpay->send_config) {
|
|
GstBuffer *subbuf;
|
|
|
|
/* strip off header */
|
|
subbuf = gst_buffer_create_sub (buffer, strip, size - strip);
|
|
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buffer);
|
|
gst_buffer_unref (buffer);
|
|
buffer = subbuf;
|
|
|
|
size = GST_BUFFER_SIZE (buffer);
|
|
data = GST_BUFFER_DATA (buffer);
|
|
}
|
|
}
|
|
|
|
/* if we need to flush, do so now */
|
|
if (flush) {
|
|
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
|
|
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
|
|
rtpmp4vpay->duration = 0;
|
|
avail = 0;
|
|
}
|
|
|
|
/* get packet length of data and see if we exceeded MTU. */
|
|
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
|
|
|
|
if (gst_basertppayload_is_filled (basepayload,
|
|
packet_len, rtpmp4vpay->duration + duration)) {
|
|
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
|
|
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
|
|
rtpmp4vpay->duration = 0;
|
|
}
|
|
|
|
/* push new data */
|
|
gst_adapter_push (rtpmp4vpay->adapter, buffer);
|
|
rtpmp4vpay->duration += duration;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpMP4VPay *rtpmp4vpay;
|
|
|
|
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_SEND_CONFIG:
|
|
rtpmp4vpay->send_config = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpMP4VPay *rtpmp4vpay;
|
|
|
|
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_SEND_CONFIG:
|
|
g_value_set_boolean (value, rtpmp4vpay->send_config);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mp4v_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmp4vpay",
|
|
GST_RANK_NONE, GST_TYPE_RTP_MP4V_PAY);
|
|
}
|