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6717c86061
For more optimised RTP packet handling: means we don't need to map the input buffer again but can just re-use the mapping the base class has already done. https://bugzilla.gnome.org/show_bug.cgi?id=750235
119 lines
3.7 KiB
C
119 lines
3.7 KiB
C
/*
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* Siren Depayloader Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpsirendepay.h"
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static GstStaticPadTemplate gst_rtp_siren_depay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 16000, " "encoding-name = (string) \"SIREN\"")
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/* This is the default, so the peer doesn't have to specify it */
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/* " "dct-length = (int) 320") */
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);
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static GstStaticPadTemplate gst_rtp_siren_depay_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 ("audio/x-siren, " "dct-length = (int) 320")
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);
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static GstBuffer *gst_rtp_siren_depay_process (GstRTPBaseDepayload *
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depayload, GstRTPBuffer * rtp);
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static gboolean gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload *
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depayload, GstCaps * caps);
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G_DEFINE_TYPE (GstRTPSirenDepay, gst_rtp_siren_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_siren_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_siren_depay_setcaps;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_siren_depay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_siren_depay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Siren packet depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts Siren audio from RTP packets",
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"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
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}
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static void
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gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay)
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{
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}
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static gboolean
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gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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gboolean ret;
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srccaps = gst_caps_new_simple ("audio/x-siren",
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"dct-length", G_TYPE_INT, 320, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
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gst_caps_unref (srccaps);
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/* always fixed clock rate of 16000 */
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depayload->clock_rate = 16000;
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return ret;
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}
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static GstBuffer *
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gst_rtp_siren_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp)
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{
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GstBuffer *outbuf;
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
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return outbuf;
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}
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gboolean
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gst_rtp_siren_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpsirendepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_DEPAY);
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}
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