1 /* GStreamer
2 * Copyright (C) <2007> Nokia Corporation
3 * Copyright (C) <2007> Collabora Ltd
4 * @author: Olivier Crete <olivier.crete@collabora.co.uk>
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 <string.h>
27 #include <gst/rtp/gstrtpbuffer.h>
28 #include <gst/base/gstadapter.h>
29 #include <gst/audio/audio.h>
30
31 #include "gstrtpg723pay.h"
32 #include "gstrtputils.h"
33
34 #define G723_FRAME_DURATION (30 * GST_MSECOND)
35
36 static gboolean gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload,
37 GstCaps * caps);
38 static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload *
39 payload, GstBuffer * buf);
40
41 static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
42 GST_STATIC_PAD_TEMPLATE ("sink",
43 GST_PAD_SINK,
44 GST_PAD_ALWAYS,
45 GST_STATIC_CAPS ("audio/G723, " /* according to RFC 3551 */
46 "channels = (int) 1, " "rate = (int) 8000")
47 );
48
49 static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
50 GST_STATIC_PAD_TEMPLATE ("src",
51 GST_PAD_SRC,
52 GST_PAD_ALWAYS,
53 GST_STATIC_CAPS ("application/x-rtp, "
54 "media = (string) \"audio\", "
55 "payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
56 "clock-rate = (int) 8000, "
57 "encoding-name = (string) \"G723\"; "
58 "application/x-rtp, "
59 "media = (string) \"audio\", "
60 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
61 "clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
62 );
63
64 static void gst_rtp_g723_pay_finalize (GObject * object);
65
66 static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
67 GstStateChange transition);
68
69 #define gst_rtp_g723_pay_parent_class parent_class
70 G_DEFINE_TYPE (GstRTPG723Pay, gst_rtp_g723_pay, GST_TYPE_RTP_BASE_PAYLOAD);
71
72 static void
gst_rtp_g723_pay_class_init(GstRTPG723PayClass * klass)73 gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
74 {
75 GObjectClass *gobject_class;
76 GstElementClass *gstelement_class;
77 GstRTPBasePayloadClass *payload_class;
78
79 gobject_class = (GObjectClass *) klass;
80 gstelement_class = (GstElementClass *) klass;
81 payload_class = (GstRTPBasePayloadClass *) klass;
82
83 gobject_class->finalize = gst_rtp_g723_pay_finalize;
84
85 gstelement_class->change_state = gst_rtp_g723_pay_change_state;
86
87 gst_element_class_add_static_pad_template (gstelement_class,
88 &gst_rtp_g723_pay_sink_template);
89 gst_element_class_add_static_pad_template (gstelement_class,
90 &gst_rtp_g723_pay_src_template);
91
92 gst_element_class_set_static_metadata (gstelement_class,
93 "RTP G.723 payloader", "Codec/Payloader/Network/RTP",
94 "Packetize G.723 audio into RTP packets",
95 "Wim Taymans <wim.taymans@gmail.com>");
96
97 payload_class->set_caps = gst_rtp_g723_pay_set_caps;
98 payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
99 }
100
101 static void
gst_rtp_g723_pay_init(GstRTPG723Pay * pay)102 gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
103 {
104 GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
105
106 pay->adapter = gst_adapter_new ();
107
108 payload->pt = GST_RTP_PAYLOAD_G723;
109 }
110
111 static void
gst_rtp_g723_pay_finalize(GObject * object)112 gst_rtp_g723_pay_finalize (GObject * object)
113 {
114 GstRTPG723Pay *pay;
115
116 pay = GST_RTP_G723_PAY (object);
117
118 g_object_unref (pay->adapter);
119 pay->adapter = NULL;
120
121 G_OBJECT_CLASS (parent_class)->finalize (object);
122 }
123
124
125 static gboolean
gst_rtp_g723_pay_set_caps(GstRTPBasePayload * payload,GstCaps * caps)126 gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
127 {
128 gboolean res;
129
130 gst_rtp_base_payload_set_options (payload, "audio",
131 payload->pt != GST_RTP_PAYLOAD_G723, "G723", 8000);
132 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
133
134 return res;
135 }
136
137 static GstFlowReturn
gst_rtp_g723_pay_flush(GstRTPG723Pay * pay)138 gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
139 {
140 GstBuffer *outbuf, *payload_buf;
141 GstFlowReturn ret;
142 guint avail;
143 GstRTPBuffer rtp = { NULL };
144
145 avail = gst_adapter_available (pay->adapter);
146
147 outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
148
149 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
150
151 GST_BUFFER_PTS (outbuf) = pay->timestamp;
152 GST_BUFFER_DURATION (outbuf) = pay->duration;
153
154 /* copy G723 data as payload */
155 payload_buf = gst_adapter_take_buffer_fast (pay->adapter, avail);
156
157 pay->timestamp = GST_CLOCK_TIME_NONE;
158 pay->duration = 0;
159
160 /* set discont and marker */
161 if (pay->discont) {
162 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
163 gst_rtp_buffer_set_marker (&rtp, TRUE);
164 pay->discont = FALSE;
165 }
166 gst_rtp_buffer_unmap (&rtp);
167 gst_rtp_copy_audio_meta (pay, outbuf, payload_buf);
168
169 outbuf = gst_buffer_append (outbuf, payload_buf);
170
171 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
172
173 return ret;
174 }
175
176 /* 00 high-rate speech (6.3 kb/s) 24
177 * 01 low-rate speech (5.3 kb/s) 20
178 * 10 SID frame 4
179 * 11 reserved 0 */
180 static const guint size_tab[4] = {
181 24, 20, 4, 0
182 };
183
184 static GstFlowReturn
gst_rtp_g723_pay_handle_buffer(GstRTPBasePayload * payload,GstBuffer * buf)185 gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
186 {
187 GstFlowReturn ret = GST_FLOW_OK;
188 GstMapInfo map;
189 guint8 HDR;
190 GstRTPG723Pay *pay;
191 GstClockTime packet_dur, timestamp;
192 guint payload_len, packet_len;
193
194 pay = GST_RTP_G723_PAY (payload);
195
196 gst_buffer_map (buf, &map, GST_MAP_READ);
197 timestamp = GST_BUFFER_PTS (buf);
198
199 if (GST_BUFFER_IS_DISCONT (buf)) {
200 /* flush everything on discont */
201 gst_adapter_clear (pay->adapter);
202 pay->timestamp = GST_CLOCK_TIME_NONE;
203 pay->duration = 0;
204 pay->discont = TRUE;
205 }
206
207 /* should be one of these sizes */
208 if (map.size != 4 && map.size != 20 && map.size != 24)
209 goto invalid_size;
210
211 /* check size by looking at the header bits */
212 HDR = map.data[0] & 0x3;
213 if (size_tab[HDR] != map.size)
214 goto wrong_size;
215
216 /* calculate packet size and duration */
217 payload_len = gst_adapter_available (pay->adapter) + map.size;
218 packet_dur = pay->duration + G723_FRAME_DURATION;
219 packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
220
221 if (gst_rtp_base_payload_is_filled (payload, packet_len, packet_dur)) {
222 /* size or duration would overflow the packet, flush the queued data */
223 ret = gst_rtp_g723_pay_flush (pay);
224 }
225
226 /* update timestamp, we keep the timestamp for the first packet in the adapter
227 * but are able to calculate it from next packets. */
228 if (timestamp != GST_CLOCK_TIME_NONE && pay->timestamp == GST_CLOCK_TIME_NONE) {
229 if (timestamp > pay->duration)
230 pay->timestamp = timestamp - pay->duration;
231 else
232 pay->timestamp = 0;
233 }
234 gst_buffer_unmap (buf, &map);
235
236 /* add packet to the queue */
237 gst_adapter_push (pay->adapter, buf);
238 pay->duration = packet_dur;
239
240 /* check if we can flush now */
241 if (pay->duration >= payload->min_ptime) {
242 ret = gst_rtp_g723_pay_flush (pay);
243 }
244
245 return ret;
246
247 /* WARNINGS */
248 invalid_size:
249 {
250 GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
251 ("Invalid input buffer size"),
252 ("Input size should be 4, 20 or 24, got %" G_GSIZE_FORMAT, map.size));
253 gst_buffer_unmap (buf, &map);
254 gst_buffer_unref (buf);
255 return GST_FLOW_OK;
256 }
257 wrong_size:
258 {
259 GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
260 ("Wrong input buffer size"),
261 ("Expected input buffer size %u but got %" G_GSIZE_FORMAT,
262 size_tab[HDR], map.size));
263 gst_buffer_unmap (buf, &map);
264 gst_buffer_unref (buf);
265 return GST_FLOW_OK;
266 }
267 }
268
269 static GstStateChangeReturn
gst_rtp_g723_pay_change_state(GstElement * element,GstStateChange transition)270 gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
271 {
272 GstStateChangeReturn ret;
273 GstRTPG723Pay *pay;
274
275 pay = GST_RTP_G723_PAY (element);
276
277 switch (transition) {
278 case GST_STATE_CHANGE_READY_TO_PAUSED:
279 gst_adapter_clear (pay->adapter);
280 pay->timestamp = GST_CLOCK_TIME_NONE;
281 pay->duration = 0;
282 pay->discont = TRUE;
283 break;
284 default:
285 break;
286 }
287
288 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
289
290 switch (transition) {
291 case GST_STATE_CHANGE_PAUSED_TO_READY:
292 gst_adapter_clear (pay->adapter);
293 break;
294 default:
295 break;
296 }
297
298 return ret;
299 }
300
301 /*Plugin init functions*/
302 gboolean
gst_rtp_g723_pay_plugin_init(GstPlugin * plugin)303 gst_rtp_g723_pay_plugin_init (GstPlugin * plugin)
304 {
305 return gst_element_register (plugin, "rtpg723pay", GST_RANK_SECONDARY,
306 gst_rtp_g723_pay_get_type ());
307 }
308