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 /* elementfactory information */
30 static GstElementDetails gst_rtpsirenpay_details = {
31 "RTP Payloader for Siren Audio",
32 "Codec/Payloader/Network",
33 "Packetize Siren audio streams into RTP packets",
34 "Youness Alaoui <kakaroto@kakaroto.homelinux.net>"
37 GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
38 #define GST_CAT_DEFAULT (rtpsirenpay_debug)
40 static GstStaticPadTemplate gst_rtpsirenpay_sink_template =
41 GST_STATIC_PAD_TEMPLATE ("sink",
44 GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
47 static GstStaticPadTemplate gst_rtpsirenpay_src_template =
48 GST_STATIC_PAD_TEMPLATE ("src",
51 GST_STATIC_CAPS ("application/x-rtp, "
52 "media = (string) \"audio\", "
53 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
54 "clock-rate = (int) 16000, "
55 "encoding-name = (string) \"SIREN\", "
56 "bitrate = (string) \"16000\", " "dct-length = (int) 320")
59 static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,
62 GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload,
63 GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
66 gst_rtpsirenpay_base_init (gpointer klass)
68 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
70 gst_element_class_add_pad_template (element_class,
71 gst_static_pad_template_get (&gst_rtpsirenpay_sink_template));
72 gst_element_class_add_pad_template (element_class,
73 gst_static_pad_template_get (&gst_rtpsirenpay_src_template));
74 gst_element_class_set_details (element_class, &gst_rtpsirenpay_details);
78 gst_rtpsirenpay_class_init (GstRTPSirenPayClass * klass)
80 GstBaseRTPPayloadClass *gstbasertppayload_class;
82 gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
84 parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
86 gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps;
88 GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
89 "siren audio RTP payloader");
93 gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass)
95 GstBaseRTPPayload *basertppayload;
96 GstBaseRTPAudioPayload *basertpaudiopayload;
98 basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
99 basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
101 /* we don't set the payload type, it should be set by the application using
102 * the pt property or the default 96 will be used */
103 basertppayload->clock_rate = 16000;
105 /* tell basertpaudiopayload that this is a frame based codec */
106 gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
110 gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
112 GstRTPSirenPay *rtpsirenpay;
113 GstBaseRTPAudioPayload *basertpaudiopayload;
115 GstStructure *structure;
116 const char *payload_name;
118 rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
119 basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
121 structure = gst_caps_get_structure (caps, 0);
123 gst_structure_get_int (structure, "dct-length", &dct_length);
124 if (dct_length != 320)
127 payload_name = gst_structure_get_name (structure);
128 if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
131 gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
133 /* set options for this frame based audio codec */
134 gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
136 return gst_basertppayload_set_outcaps (basertppayload, NULL);
141 GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
147 GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
154 gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
156 return gst_element_register (plugin, "rtpsirenpay",
157 GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);