2 * Siren Payloader Gst Element
4 * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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.
26 #include "gstrtpsirenpay.h"
27 #include <gst/rtp/gstrtpbuffer.h>
29 GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
30 #define GST_CAT_DEFAULT (rtpsirenpay_debug)
32 static GstStaticPadTemplate gst_rtp_siren_pay_sink_template =
33 GST_STATIC_PAD_TEMPLATE ("sink",
36 GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
39 static GstStaticPadTemplate gst_rtp_siren_pay_src_template =
40 GST_STATIC_PAD_TEMPLATE ("src",
43 GST_STATIC_CAPS ("application/x-rtp, "
44 "media = (string) \"audio\", "
45 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
46 "clock-rate = (int) 16000, "
47 "encoding-name = (string) \"SIREN\", "
48 "bitrate = (string) \"16000\", " "dct-length = (int) 320")
51 static gboolean gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * payload,
54 GST_BOILERPLATE (GstRTPSirenPay, gst_rtp_siren_pay, GstBaseRTPAudioPayload,
55 GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
58 gst_rtp_siren_pay_base_init (gpointer klass)
60 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
62 gst_element_class_add_pad_template (element_class,
63 gst_static_pad_template_get (&gst_rtp_siren_pay_sink_template));
64 gst_element_class_add_pad_template (element_class,
65 gst_static_pad_template_get (&gst_rtp_siren_pay_src_template));
66 gst_element_class_set_details_simple (element_class,
67 "RTP Payloader for Siren Audio", "Codec/Payloader/Network",
68 "Packetize Siren audio streams into RTP packets",
69 "Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
73 gst_rtp_siren_pay_class_init (GstRTPSirenPayClass * klass)
75 GstBaseRTPPayloadClass *gstbasertppayload_class;
77 gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
79 gstbasertppayload_class->set_caps = gst_rtp_siren_pay_setcaps;
81 GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
82 "siren audio RTP payloader");
86 gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay,
87 GstRTPSirenPayClass * klass)
89 GstBaseRTPPayload *basertppayload;
90 GstBaseRTPAudioPayload *basertpaudiopayload;
92 basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
93 basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
95 /* we don't set the payload type, it should be set by the application using
96 * the pt property or the default 96 will be used */
97 basertppayload->clock_rate = 16000;
99 /* tell basertpaudiopayload that this is a frame based codec */
100 gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
104 gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
106 GstRTPSirenPay *rtpsirenpay;
107 GstBaseRTPAudioPayload *basertpaudiopayload;
109 GstStructure *structure;
110 const char *payload_name;
112 rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
113 basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
115 structure = gst_caps_get_structure (caps, 0);
117 gst_structure_get_int (structure, "dct-length", &dct_length);
118 if (dct_length != 320)
121 payload_name = gst_structure_get_name (structure);
122 if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
125 gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
127 /* set options for this frame based audio codec */
128 gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
130 return gst_basertppayload_set_outcaps (basertppayload, NULL);
135 GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
141 GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
148 gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
150 return gst_element_register (plugin, "rtpsirenpay",
151 GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);