gstreamer/gst/rtp/gstrtpopusdepay.c
Nicolas Dufresne 9e4511edf4 rtpopus: Use OPUS encoding name
Both Firefox and Chrome uses OPUS as the encoding in their SDP.
Adding this now defacto standard name remove the need for special
case in SDP parsing code.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2016-02-17 14:58:01 +00:00

120 lines
3.7 KiB
C

/*
* Opus Depayloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpopusdepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
#define GST_CAT_DEFAULT (rtpopusdepay_debug)
static GstStaticPadTemplate gst_rtp_opus_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
"clock-rate = (int) 48000, "
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
);
static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus")
);
static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
{
GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
GstElementClass *element_class;
element_class = GST_ELEMENT_CLASS (klass);
gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_depay_sink_template));
gst_element_class_set_static_metadata (element_class,
"RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
"Extracts Opus audio from RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
gstbasertpdepayload_class->process = gst_rtp_opus_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
"Opus RTP Depayloader");
}
static void
gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
{
}
static gboolean
gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
srccaps = gst_caps_new_empty_simple ("audio/x-opus");
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
GST_DEBUG_OBJECT (depayload,
"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
gst_caps_unref (srccaps);
depayload->clock_rate = 48000;
return ret;
}
static GstBuffer *
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
GstRTPBuffer rtpbuf = { NULL, };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
return outbuf;
}