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 (GstBaseRTPPayload * payload,
56 static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload *
57 payload, GstBuffer * buffer);
59 GST_BOILERPLATE (GstRtpOPUSPay, gst_rtp_opus_pay, GstBaseRTPPayload,
60 GST_TYPE_BASE_RTP_PAYLOAD);
63 gst_rtp_opus_pay_base_init (gpointer klass)
65 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
67 gst_element_class_add_pad_template (element_class,
68 gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
69 gst_element_class_add_pad_template (element_class,
70 gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
72 gst_element_class_set_details_simple (element_class,
74 "Codec/Payloader/Network/RTP",
75 "Puts Opus audio in RTP packets",
76 "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
80 gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
82 GstBaseRTPPayloadClass *gstbasertppayload_class;
84 gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
86 gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
87 gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
89 GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
90 "Opus RTP Payloader");
94 gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay, GstRtpOPUSPayClass * klass)
99 gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
104 capsstr = gst_caps_to_string (caps);
106 gst_basertppayload_set_options (payload, "audio", FALSE,
107 "X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
109 gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
117 gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload * basepayload,
121 GstClockTime timestamp;
127 size = GST_BUFFER_SIZE (buffer);
128 data = GST_BUFFER_DATA (buffer);
129 timestamp = GST_BUFFER_TIMESTAMP (buffer);
131 outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
132 payload = gst_rtp_buffer_get_payload (outbuf);
134 memcpy (payload, data, size);
136 gst_rtp_buffer_set_marker (outbuf, FALSE);
137 GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
139 return gst_basertppayload_push (basepayload, outbuf);