1 /*
2 * Siren Payloader Gst Element
3 *
4 * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
5 *
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.
10 *
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.
15 *
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., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
20 */
21
22 #ifdef HAVE_CONFIG_H
23 #include "config.h"
24 #endif
25
26 #include "gstrtpsirenpay.h"
27 #include <gst/rtp/gstrtpbuffer.h>
28
29 GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
30 #define GST_CAT_DEFAULT (rtpsirenpay_debug)
31
32 static GstStaticPadTemplate gst_rtp_siren_pay_sink_template =
33 GST_STATIC_PAD_TEMPLATE ("sink",
34 GST_PAD_SINK,
35 GST_PAD_ALWAYS,
36 GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
37 );
38
39 static GstStaticPadTemplate gst_rtp_siren_pay_src_template =
40 GST_STATIC_PAD_TEMPLATE ("src",
41 GST_PAD_SRC,
42 GST_PAD_ALWAYS,
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")
49 );
50
51 static gboolean gst_rtp_siren_pay_setcaps (GstRTPBasePayload * payload,
52 GstCaps * caps);
53
54 G_DEFINE_TYPE (GstRTPSirenPay, gst_rtp_siren_pay,
55 GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
56
57 static void
gst_rtp_siren_pay_class_init(GstRTPSirenPayClass * klass)58 gst_rtp_siren_pay_class_init (GstRTPSirenPayClass * klass)
59 {
60 GstElementClass *gstelement_class;
61 GstRTPBasePayloadClass *gstrtpbasepayload_class;
62
63 gstelement_class = (GstElementClass *) klass;
64 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
65
66 gstrtpbasepayload_class->set_caps = gst_rtp_siren_pay_setcaps;
67
68 gst_element_class_add_static_pad_template (gstelement_class,
69 &gst_rtp_siren_pay_sink_template);
70 gst_element_class_add_static_pad_template (gstelement_class,
71 &gst_rtp_siren_pay_src_template);
72 gst_element_class_set_static_metadata (gstelement_class,
73 "RTP Payloader for Siren Audio", "Codec/Payloader/Network/RTP",
74 "Packetize Siren audio streams into RTP packets",
75 "Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
76
77 GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
78 "siren audio RTP payloader");
79 }
80
81 static void
gst_rtp_siren_pay_init(GstRTPSirenPay * rtpsirenpay)82 gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay)
83 {
84 GstRTPBasePayload *rtpbasepayload;
85 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
86
87 rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpsirenpay);
88 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpsirenpay);
89
90 /* we don't set the payload type, it should be set by the application using
91 * the pt property or the default 96 will be used */
92 rtpbasepayload->clock_rate = 16000;
93
94 /* tell rtpbaseaudiopayload that this is a frame based codec */
95 gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
96 }
97
98 static gboolean
gst_rtp_siren_pay_setcaps(GstRTPBasePayload * rtpbasepayload,GstCaps * caps)99 gst_rtp_siren_pay_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
100 {
101 GstRTPSirenPay *rtpsirenpay;
102 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
103 gint dct_length;
104 GstStructure *structure;
105 const char *payload_name;
106
107 rtpsirenpay = GST_RTP_SIREN_PAY (rtpbasepayload);
108 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
109
110 structure = gst_caps_get_structure (caps, 0);
111
112 gst_structure_get_int (structure, "dct-length", &dct_length);
113 if (dct_length != 320)
114 goto wrong_dct;
115
116 payload_name = gst_structure_get_name (structure);
117 if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
118 goto wrong_caps;
119
120 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "SIREN",
121 16000);
122 /* set options for this frame based audio codec */
123 gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, 20, 40);
124
125 return gst_rtp_base_payload_set_outcaps (rtpbasepayload, NULL);
126
127 /* ERRORS */
128 wrong_dct:
129 {
130 GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
131 dct_length);
132 return FALSE;
133 }
134 wrong_caps:
135 {
136 GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
137 payload_name);
138 return FALSE;
139 }
140 }
141
142 gboolean
gst_rtp_siren_pay_plugin_init(GstPlugin * plugin)143 gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
144 {
145 return gst_element_register (plugin, "rtpsirenpay",
146 GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_PAY);
147 }
148