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120 lines
3.7 KiB
C
120 lines
3.7 KiB
C
/*
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* Opus Depayloader Gst Element
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*
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* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
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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 <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpopusdepay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
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#define GST_CAT_DEFAULT (rtpopusdepay_debug)
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static GstStaticPadTemplate gst_rtp_opus_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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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
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"clock-rate = (int) 48000, "
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"encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
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);
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static GstStaticPadTemplate gst_rtp_opus_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-opus")
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);
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static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
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GstBuffer * buf);
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static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void
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gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
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{
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GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
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GstElementClass *element_class;
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element_class = GST_ELEMENT_CLASS (klass);
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gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_opus_depay_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_opus_depay_sink_template));
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gst_element_class_set_metadata (element_class,
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"RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts Opus audio from RTP packets",
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"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
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gstbasertpdepayload_class->process = gst_rtp_opus_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
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"Opus RTP Depayloader");
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}
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static void
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gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
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{
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}
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static gboolean
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gst_rtp_opus_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_empty_simple ("audio/x-opus");
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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GST_DEBUG_OBJECT (depayload,
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"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
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gst_caps_unref (srccaps);
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depayload->clock_rate = 48000;
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return ret;
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}
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static GstBuffer *
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gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
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{
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GstBuffer *outbuf;
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GstRTPBuffer rtpbuf = { NULL, };
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gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
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outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
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gst_rtp_buffer_unmap (&rtpbuf);
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return outbuf;
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}
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