gstreamer/ext/opus/gstrtpopuspay.c
Nicolas Dufresne 54602c04f6 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
2014-11-01 10:10:27 -04:00

125 lines
3.8 KiB
C

/*
* 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) { \"OPUS\", \"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);
}