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/*
* Opus Payloader 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 <gst/rtp/gstrtpbuffer.h>
#include "gstrtpopuspay.h"
GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
#define GST_CAT_DEFAULT (rtpopuspay_debug)
static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, multistream = (boolean) FALSE")
);
static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
);
static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
{
GstRTPBasePayloadClass *gstbasertppayload_class;
GstElementClass *element_class;
gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
element_class = GST_ELEMENT_CLASS (klass);
gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
gst_element_class_set_static_metadata (element_class,
"RTP Opus payloader",
"Codec/Payloader/Network/RTP",
"Puts Opus audio in RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
"Opus RTP Payloader");
}
static void
gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
{
}
static gboolean
gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "audio", FALSE,
"X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstBuffer *outbuf;
GstClockTime pts, dts, duration;
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
outbuf = gst_buffer_append (outbuf, buffer);
GST_BUFFER_PTS (outbuf) = pts;
GST_BUFFER_DTS (outbuf) = dts;
GST_BUFFER_DURATION (outbuf) = duration;
/* Push out */
return gst_rtp_base_payload_push (basepayload, outbuf);
}
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