2 * Opus Payloader Gst Element
4 * @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
19 * Boston, MA 02111-1307, USA.
28 #include <gst/rtp/gstrtpbuffer.h>
30 #include "gstrtpopuspay.h"
32 GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
33 #define GST_CAT_DEFAULT (rtpopuspay_debug)
36 static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
37 GST_STATIC_PAD_TEMPLATE ("sink",
40 GST_STATIC_CAPS ("audio/x-opus, multistream = (boolean) FALSE")
43 static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
44 GST_STATIC_PAD_TEMPLATE ("src",
47 GST_STATIC_CAPS ("application/x-rtp, "
48 "media = (string) \"audio\", "
49 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
50 "clock-rate = (int) 48000, "
51 "encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
54 static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
56 static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
57 payload, GstBuffer * buffer);
59 G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
62 gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
64 GstRTPBasePayloadClass *gstbasertppayload_class;
65 GstElementClass *element_class;
67 gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
68 element_class = GST_ELEMENT_CLASS (klass);
70 gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
71 gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
73 gst_element_class_add_pad_template (element_class,
74 gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
75 gst_element_class_add_pad_template (element_class,
76 gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
78 gst_element_class_set_details_simple (element_class,
80 "Codec/Payloader/Network/RTP",
81 "Puts Opus audio in RTP packets",
82 "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
84 GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
85 "Opus RTP Payloader");
89 gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
94 gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
99 capsstr = gst_caps_to_string (caps);
101 gst_rtp_base_payload_set_options (payload, "audio", FALSE,
102 "X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
104 gst_rtp_base_payload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
112 gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
115 GstRTPBuffer rtpbuf = { NULL, };
119 /* Copy data and timestamp to a new output buffer
120 * FIXME : Don't we have a convenience function for this ? */
121 gst_buffer_map (buffer, &map, GST_MAP_READ);
122 outbuf = gst_rtp_buffer_new_copy_data (map.data, map.size);
123 GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buffer);
125 /* Unmap and free input buffer */
126 gst_buffer_unmap (buffer, &map);
127 gst_buffer_unref (buffer);
129 /* Remove marker from RTP buffer */
130 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtpbuf);
131 gst_rtp_buffer_set_marker (&rtpbuf, FALSE);
132 gst_rtp_buffer_unmap (&rtpbuf);
135 return gst_rtp_base_payload_push (basepayload, outbuf);