1 /* GStreamer
2 * Copyright (C) 2020 Collabora Ltd.
3 * Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
4 *
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
9 *
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
14 *
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
19 */
20
21 /**
22 * SECTION:element-rtpisacpay
23 * @title: rtpisacpay
24 * @short_description: iSAC RTP Payloader
25 *
26 * Since: 1.20
27 *
28 */
29
30 #ifdef HAVE_CONFIG_H
31 #include "config.h"
32 #endif
33
34 #include <gst/rtp/gstrtpbuffer.h>
35
36 #include "gstrtpelements.h"
37 #include "gstrtpisacpay.h"
38 #include "gstrtputils.h"
39
40 GST_DEBUG_CATEGORY_STATIC (rtpisacpay_debug);
41 #define GST_CAT_DEFAULT (rtpisacpay_debug)
42
43 static GstStaticPadTemplate gst_rtp_isac_pay_sink_template =
44 GST_STATIC_PAD_TEMPLATE ("sink",
45 GST_PAD_SINK,
46 GST_PAD_ALWAYS,
47 GST_STATIC_CAPS ("audio/isac, "
48 "rate = (int) { 16000, 32000 }, " "channels = (int) 1")
49 );
50
51 static GstStaticPadTemplate gst_rtp_isac_pay_src_template =
52 GST_STATIC_PAD_TEMPLATE ("src",
53 GST_PAD_SRC,
54 GST_PAD_ALWAYS,
55 GST_STATIC_CAPS ("application/x-rtp, "
56 "media = (string) \"audio\", "
57 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
58 "clock-rate = (int) { 16000, 32000 }, "
59 "encoding-name = (string) \"ISAC\", "
60 "encoding-params = (string) \"1\"")
61 );
62
63 struct _GstRtpIsacPay
64 {
65 /*< private > */
66 GstRTPBasePayload parent;
67 };
68
69 #define gst_rtp_isac_pay_parent_class parent_class
70 G_DEFINE_TYPE (GstRtpIsacPay, gst_rtp_isac_pay, GST_TYPE_RTP_BASE_PAYLOAD);
71 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpisacpay, "rtpisacpay",
72 GST_RANK_SECONDARY, GST_TYPE_RTP_ISAC_PAY, rtp_element_init (plugin));
73
74 static GstCaps *
gst_rtp_isac_pay_getcaps(GstRTPBasePayload * payload,GstPad * pad,GstCaps * filter)75 gst_rtp_isac_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
76 GstCaps * filter)
77 {
78 GstCaps *otherpadcaps;
79 GstCaps *caps;
80
81 otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
82 caps = gst_pad_get_pad_template_caps (pad);
83
84 if (otherpadcaps) {
85 if (!gst_caps_is_empty (otherpadcaps)) {
86 GstStructure *ps;
87 GstStructure *s;
88 const GValue *v;
89
90 ps = gst_caps_get_structure (otherpadcaps, 0);
91 caps = gst_caps_make_writable (caps);
92 s = gst_caps_get_structure (caps, 0);
93
94 v = gst_structure_get_value (ps, "clock-rate");
95 if (v)
96 gst_structure_set_value (s, "rate", v);
97 }
98 gst_caps_unref (otherpadcaps);
99 }
100
101 if (filter) {
102 GstCaps *tcaps = caps;
103
104 caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
105 gst_caps_unref (tcaps);
106 }
107
108 GST_DEBUG_OBJECT (payload, "%" GST_PTR_FORMAT, caps);
109
110 return caps;
111 }
112
113 static gboolean
gst_rtp_isac_pay_setcaps(GstRTPBasePayload * payload,GstCaps * caps)114 gst_rtp_isac_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
115 {
116 GstStructure *s;
117 gint rate;
118
119 GST_DEBUG_OBJECT (payload, "%" GST_PTR_FORMAT, caps);
120
121 s = gst_caps_get_structure (caps, 0);
122 if (!gst_structure_get_int (s, "rate", &rate)) {
123 GST_ERROR_OBJECT (payload, "Missing 'rate' in caps");
124 return FALSE;
125 }
126
127 gst_rtp_base_payload_set_options (payload, "audio", TRUE, "ISAC", rate);
128
129 return gst_rtp_base_payload_set_outcaps (payload, NULL);
130 }
131
132 static GstFlowReturn
gst_rtp_isac_pay_handle_buffer(GstRTPBasePayload * basepayload,GstBuffer * buffer)133 gst_rtp_isac_pay_handle_buffer (GstRTPBasePayload * basepayload,
134 GstBuffer * buffer)
135 {
136 GstBuffer *outbuf;
137 GstClockTime pts, dts, duration;
138
139 pts = GST_BUFFER_PTS (buffer);
140 dts = GST_BUFFER_DTS (buffer);
141 duration = GST_BUFFER_DURATION (buffer);
142
143 outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
144
145 gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
146
147 outbuf = gst_buffer_append (outbuf, buffer);
148
149 GST_BUFFER_PTS (outbuf) = pts;
150 GST_BUFFER_DTS (outbuf) = dts;
151 GST_BUFFER_DURATION (outbuf) = duration;
152
153 return gst_rtp_base_payload_push (basepayload, outbuf);
154 }
155
156 static void
gst_rtp_isac_pay_class_init(GstRtpIsacPayClass * klass)157 gst_rtp_isac_pay_class_init (GstRtpIsacPayClass * klass)
158 {
159 GstElementClass *gstelement_class = (GstElementClass *) klass;
160 GstRTPBasePayloadClass *payload_class = (GstRTPBasePayloadClass *) klass;
161
162 payload_class->get_caps = gst_rtp_isac_pay_getcaps;
163 payload_class->set_caps = gst_rtp_isac_pay_setcaps;
164 payload_class->handle_buffer = gst_rtp_isac_pay_handle_buffer;
165
166 gst_element_class_add_static_pad_template (gstelement_class,
167 &gst_rtp_isac_pay_sink_template);
168 gst_element_class_add_static_pad_template (gstelement_class,
169 &gst_rtp_isac_pay_src_template);
170
171 gst_element_class_set_static_metadata (gstelement_class,
172 "RTP iSAC payloader", "Codec/Payloader/Network/RTP",
173 "Payload-encodes iSAC audio into a RTP packet",
174 "Guillaume Desmottes <guillaume.desmottes@collabora.com>");
175
176 GST_DEBUG_CATEGORY_INIT (rtpisacpay_debug, "rtpisacpay", 0,
177 "iSAC RTP Payloader");
178 }
179
180 static void
gst_rtp_isac_pay_init(GstRtpIsacPay * rtpisacpay)181 gst_rtp_isac_pay_init (GstRtpIsacPay * rtpisacpay)
182 {
183 }
184