2 * Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
25 #include <gst/rtp/gstrtpbuffer.h>
26 #include "gstrtpelements.h"
27 #include "gstrtpilbcpay.h"
29 GST_DEBUG_CATEGORY_STATIC (rtpilbcpay_debug);
30 #define GST_CAT_DEFAULT (rtpilbcpay_debug)
32 static GstStaticPadTemplate gst_rtp_ilbc_pay_sink_template =
33 GST_STATIC_PAD_TEMPLATE ("sink",
36 GST_STATIC_CAPS ("audio/x-iLBC, " "mode = (int) {20, 30}")
39 static GstStaticPadTemplate gst_rtp_ilbc_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) 8000, "
47 "encoding-name = (string) \"ILBC\", "
48 "mode = (string) { \"20\", \"30\" }")
52 static GstCaps *gst_rtp_ilbc_pay_sink_getcaps (GstRTPBasePayload * payload,
53 GstPad * pad, GstCaps * filter);
54 static gboolean gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * payload,
57 #define gst_rtp_ilbc_pay_parent_class parent_class
58 G_DEFINE_TYPE (GstRTPILBCPay, gst_rtp_ilbc_pay,
59 GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
60 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpilbcpay, "rtpilbcpay",
61 GST_RANK_SECONDARY, GST_TYPE_RTP_ILBC_PAY, rtp_element_init (plugin));
64 gst_rtp_ilbc_pay_class_init (GstRTPILBCPayClass * klass)
66 GstElementClass *gstelement_class;
67 GstRTPBasePayloadClass *gstrtpbasepayload_class;
69 GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0,
70 "iLBC audio RTP payloader");
72 gstelement_class = (GstElementClass *) klass;
73 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
75 gst_element_class_add_static_pad_template (gstelement_class,
76 &gst_rtp_ilbc_pay_sink_template);
77 gst_element_class_add_static_pad_template (gstelement_class,
78 &gst_rtp_ilbc_pay_src_template);
80 gst_element_class_set_static_metadata (gstelement_class, "RTP iLBC Payloader",
81 "Codec/Payloader/Network/RTP",
82 "Packetize iLBC audio streams into RTP packets",
83 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
85 gstrtpbasepayload_class->set_caps = gst_rtp_ilbc_pay_sink_setcaps;
86 gstrtpbasepayload_class->get_caps = gst_rtp_ilbc_pay_sink_getcaps;
90 gst_rtp_ilbc_pay_init (GstRTPILBCPay * rtpilbcpay)
92 GstRTPBasePayload *rtpbasepayload;
93 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
95 rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpilbcpay);
96 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpilbcpay);
98 /* we don't set the payload type, it should be set by the application using
99 * the pt property or the default 96 will be used */
100 rtpbasepayload->clock_rate = 8000;
102 rtpilbcpay->mode = -1;
104 /* tell rtpbaseaudiopayload that this is a frame based codec */
105 gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
109 gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload,
112 GstRTPILBCPay *rtpilbcpay;
113 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
117 GstStructure *structure;
118 const char *payload_name;
120 rtpilbcpay = GST_RTP_ILBC_PAY (rtpbasepayload);
121 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
123 structure = gst_caps_get_structure (caps, 0);
125 payload_name = gst_structure_get_name (structure);
126 if (g_ascii_strcasecmp ("audio/x-iLBC", payload_name))
129 if (!gst_structure_get_int (structure, "mode", &mode))
132 if (mode != 20 && mode != 30)
135 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "ILBC",
137 /* set options for this frame based audio codec */
138 gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
139 mode, mode == 30 ? 50 : 38);
141 mode_str = g_strdup_printf ("%d", mode);
143 gst_rtp_base_payload_set_outcaps (rtpbasepayload, "mode", G_TYPE_STRING,
147 if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1)
150 rtpilbcpay->mode = mode;
157 GST_ERROR_OBJECT (rtpilbcpay, "expected audio/x-iLBC, received %s",
163 GST_ERROR_OBJECT (rtpilbcpay, "did not receive a mode");
168 GST_ERROR_OBJECT (rtpilbcpay, "mode must be 20 or 30, received %d", mode);
173 GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! "
174 "Mode cannot change while streaming", rtpilbcpay->mode, mode);
179 /* we return the padtemplate caps with the mode field fixated to a value if we
182 gst_rtp_ilbc_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
185 GstCaps *otherpadcaps;
188 otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
189 caps = gst_pad_get_pad_template_caps (pad);
192 if (!gst_caps_is_empty (otherpadcaps)) {
193 GstStructure *structure;
194 const gchar *mode_str;
197 structure = gst_caps_get_structure (otherpadcaps, 0);
199 /* parse mode, if we can */
200 mode_str = gst_structure_get_string (structure, "mode");
202 mode = strtol (mode_str, NULL, 10);
203 if (mode == 20 || mode == 30) {
204 caps = gst_caps_make_writable (caps);
205 structure = gst_caps_get_structure (caps, 0);
206 gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
210 gst_caps_unref (otherpadcaps);
216 GST_DEBUG_OBJECT (rtppayload, "Intersect %" GST_PTR_FORMAT " and filter %"
217 GST_PTR_FORMAT, caps, filter);
218 tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
219 gst_caps_unref (caps);