2 * Copyright (C) <2007> Nokia Corporation
3 * Copyright (C) <2007> Collabora Ltd
4 * @author: Olivier Crete <olivier.crete@collabora.co.uk>
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., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
27 #include <gst/rtp/gstrtpbuffer.h>
28 #include <gst/base/gstadapter.h>
29 #include <gst/audio/audio.h>
31 #include "gstrtpg723pay.h"
32 #include "gstrtputils.h"
34 #define G723_FRAME_DURATION (30 * GST_MSECOND)
36 static gboolean gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload,
38 static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload *
39 payload, GstBuffer * buf);
41 static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
42 GST_STATIC_PAD_TEMPLATE ("sink",
45 GST_STATIC_CAPS ("audio/G723, " /* according to RFC 3551 */
46 "channels = (int) 1, " "rate = (int) 8000")
49 static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
50 GST_STATIC_PAD_TEMPLATE ("src",
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\"; "
59 "media = (string) \"audio\", "
60 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
61 "clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
64 static void gst_rtp_g723_pay_finalize (GObject * object);
66 static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
67 GstStateChange transition);
69 #define gst_rtp_g723_pay_parent_class parent_class
70 G_DEFINE_TYPE (GstRTPG723Pay, gst_rtp_g723_pay, GST_TYPE_RTP_BASE_PAYLOAD);
73 gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
75 GObjectClass *gobject_class;
76 GstElementClass *gstelement_class;
77 GstRTPBasePayloadClass *payload_class;
79 gobject_class = (GObjectClass *) klass;
80 gstelement_class = (GstElementClass *) klass;
81 payload_class = (GstRTPBasePayloadClass *) klass;
83 gobject_class->finalize = gst_rtp_g723_pay_finalize;
85 gstelement_class->change_state = gst_rtp_g723_pay_change_state;
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);
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>");
97 payload_class->set_caps = gst_rtp_g723_pay_set_caps;
98 payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
102 gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
104 GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
106 pay->adapter = gst_adapter_new ();
108 payload->pt = GST_RTP_PAYLOAD_G723;
112 gst_rtp_g723_pay_finalize (GObject * object)
116 pay = GST_RTP_G723_PAY (object);
118 g_object_unref (pay->adapter);
121 G_OBJECT_CLASS (parent_class)->finalize (object);
126 gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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);
138 gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
140 GstBuffer *outbuf, *payload_buf;
143 GstRTPBuffer rtp = { NULL };
145 avail = gst_adapter_available (pay->adapter);
147 outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
149 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
151 GST_BUFFER_PTS (outbuf) = pay->timestamp;
152 GST_BUFFER_DURATION (outbuf) = pay->duration;
154 /* copy G723 data as payload */
155 payload_buf = gst_adapter_take_buffer_fast (pay->adapter, avail);
157 pay->timestamp = GST_CLOCK_TIME_NONE;
160 /* set discont and marker */
162 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
163 gst_rtp_buffer_set_marker (&rtp, TRUE);
164 pay->discont = FALSE;
166 gst_rtp_buffer_unmap (&rtp);
167 gst_rtp_copy_audio_meta (pay, outbuf, payload_buf);
169 outbuf = gst_buffer_append (outbuf, payload_buf);
171 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
176 /* 00 high-rate speech (6.3 kb/s) 24
177 * 01 low-rate speech (5.3 kb/s) 20
180 static const guint size_tab[4] = {
185 gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
187 GstFlowReturn ret = GST_FLOW_OK;
191 GstClockTime packet_dur, timestamp;
192 guint payload_len, packet_len;
194 pay = GST_RTP_G723_PAY (payload);
196 gst_buffer_map (buf, &map, GST_MAP_READ);
197 timestamp = GST_BUFFER_PTS (buf);
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;
207 /* should be one of these sizes */
208 if (map.size != 4 && map.size != 20 && map.size != 24)
211 /* check size by looking at the header bits */
212 HDR = map.data[0] & 0x3;
213 if (size_tab[HDR] != map.size)
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);
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);
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;
234 gst_buffer_unmap (buf, &map);
236 /* add packet to the queue */
237 gst_adapter_push (pay->adapter, buf);
238 pay->duration = packet_dur;
240 /* check if we can flush now */
241 if (pay->duration >= payload->min_ptime) {
242 ret = gst_rtp_g723_pay_flush (pay);
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);
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);
269 static GstStateChangeReturn
270 gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
272 GstStateChangeReturn ret;
275 pay = GST_RTP_G723_PAY (element);
277 switch (transition) {
278 case GST_STATE_CHANGE_READY_TO_PAUSED:
279 gst_adapter_clear (pay->adapter);
280 pay->timestamp = GST_CLOCK_TIME_NONE;
288 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
290 switch (transition) {
291 case GST_STATE_CHANGE_PAUSED_TO_READY:
292 gst_adapter_clear (pay->adapter);
301 /*Plugin init functions*/
303 gst_rtp_g723_pay_plugin_init (GstPlugin * plugin)
305 return gst_element_register (plugin, "rtpg723pay", GST_RANK_SECONDARY,
306 gst_rtp_g723_pay_get_type ());