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 "gstrtpelements.h"
32 #include "gstrtpg723pay.h"
33 #include "gstrtputils.h"
35 #define G723_FRAME_DURATION (30 * GST_MSECOND)
37 static gboolean gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload,
39 static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload *
40 payload, GstBuffer * buf);
42 static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
43 GST_STATIC_PAD_TEMPLATE ("sink",
46 GST_STATIC_CAPS ("audio/G723, " /* according to RFC 3551 */
47 "channels = (int) 1, " "rate = (int) 8000")
50 static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
51 GST_STATIC_PAD_TEMPLATE ("src",
54 GST_STATIC_CAPS ("application/x-rtp, "
55 "media = (string) \"audio\", "
56 "payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
57 "clock-rate = (int) 8000, "
58 "encoding-name = (string) \"G723\"; "
60 "media = (string) \"audio\", "
61 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
62 "clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
65 static void gst_rtp_g723_pay_finalize (GObject * object);
67 static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
68 GstStateChange transition);
70 #define gst_rtp_g723_pay_parent_class parent_class
71 G_DEFINE_TYPE (GstRTPG723Pay, gst_rtp_g723_pay, GST_TYPE_RTP_BASE_PAYLOAD);
72 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg723pay, "rtpg723pay",
73 GST_RANK_SECONDARY, GST_TYPE_RTP_G723_PAY, rtp_element_init (plugin));
76 gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
78 GObjectClass *gobject_class;
79 GstElementClass *gstelement_class;
80 GstRTPBasePayloadClass *payload_class;
82 gobject_class = (GObjectClass *) klass;
83 gstelement_class = (GstElementClass *) klass;
84 payload_class = (GstRTPBasePayloadClass *) klass;
86 gobject_class->finalize = gst_rtp_g723_pay_finalize;
88 gstelement_class->change_state = gst_rtp_g723_pay_change_state;
90 gst_element_class_add_static_pad_template (gstelement_class,
91 &gst_rtp_g723_pay_sink_template);
92 gst_element_class_add_static_pad_template (gstelement_class,
93 &gst_rtp_g723_pay_src_template);
95 gst_element_class_set_static_metadata (gstelement_class,
96 "RTP G.723 payloader", "Codec/Payloader/Network/RTP",
97 "Packetize G.723 audio into RTP packets",
98 "Wim Taymans <wim.taymans@gmail.com>");
100 payload_class->set_caps = gst_rtp_g723_pay_set_caps;
101 payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
105 gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
107 GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
109 pay->adapter = gst_adapter_new ();
111 payload->pt = GST_RTP_PAYLOAD_G723;
115 gst_rtp_g723_pay_finalize (GObject * object)
119 pay = GST_RTP_G723_PAY (object);
121 g_object_unref (pay->adapter);
124 G_OBJECT_CLASS (parent_class)->finalize (object);
129 gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
133 gst_rtp_base_payload_set_options (payload, "audio",
134 payload->pt != GST_RTP_PAYLOAD_G723, "G723", 8000);
135 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
141 gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
143 GstBuffer *outbuf, *payload_buf;
146 GstRTPBuffer rtp = { NULL };
148 avail = gst_adapter_available (pay->adapter);
151 gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
153 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
155 GST_BUFFER_PTS (outbuf) = pay->timestamp;
156 GST_BUFFER_DURATION (outbuf) = pay->duration;
158 /* copy G723 data as payload */
159 payload_buf = gst_adapter_take_buffer_fast (pay->adapter, avail);
161 pay->timestamp = GST_CLOCK_TIME_NONE;
164 /* set discont and marker */
166 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
167 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
168 gst_rtp_buffer_set_marker (&rtp, TRUE);
169 pay->discont = FALSE;
171 gst_rtp_buffer_unmap (&rtp);
172 gst_rtp_copy_audio_meta (pay, outbuf, payload_buf);
174 outbuf = gst_buffer_append (outbuf, payload_buf);
176 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
181 /* 00 high-rate speech (6.3 kb/s) 24
182 * 01 low-rate speech (5.3 kb/s) 20
185 static const guint size_tab[4] = {
190 gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
192 GstFlowReturn ret = GST_FLOW_OK;
196 GstClockTime packet_dur, timestamp;
197 guint payload_len, packet_len;
199 pay = GST_RTP_G723_PAY (payload);
201 gst_buffer_map (buf, &map, GST_MAP_READ);
202 timestamp = GST_BUFFER_PTS (buf);
204 if (GST_BUFFER_IS_DISCONT (buf)) {
205 /* flush everything on discont */
206 gst_adapter_clear (pay->adapter);
207 pay->timestamp = GST_CLOCK_TIME_NONE;
212 /* should be one of these sizes */
213 if (map.size != 4 && map.size != 20 && map.size != 24)
216 /* check size by looking at the header bits */
217 HDR = map.data[0] & 0x3;
218 if (size_tab[HDR] != map.size)
221 /* calculate packet size and duration */
222 payload_len = gst_adapter_available (pay->adapter) + map.size;
223 packet_dur = pay->duration + G723_FRAME_DURATION;
224 packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
226 if (gst_rtp_base_payload_is_filled (payload, packet_len, packet_dur)) {
227 /* size or duration would overflow the packet, flush the queued data */
228 ret = gst_rtp_g723_pay_flush (pay);
231 /* update timestamp, we keep the timestamp for the first packet in the adapter
232 * but are able to calculate it from next packets. */
233 if (timestamp != GST_CLOCK_TIME_NONE && pay->timestamp == GST_CLOCK_TIME_NONE) {
234 if (timestamp > pay->duration)
235 pay->timestamp = timestamp - pay->duration;
239 gst_buffer_unmap (buf, &map);
241 /* add packet to the queue */
242 gst_adapter_push (pay->adapter, buf);
243 pay->duration = packet_dur;
245 /* check if we can flush now */
246 if (pay->duration >= payload->min_ptime) {
247 ret = gst_rtp_g723_pay_flush (pay);
255 GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
256 ("Invalid input buffer size"),
257 ("Input size should be 4, 20 or 24, got %" G_GSIZE_FORMAT, map.size));
258 gst_buffer_unmap (buf, &map);
259 gst_buffer_unref (buf);
264 GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
265 ("Wrong input buffer size"),
266 ("Expected input buffer size %u but got %" G_GSIZE_FORMAT,
267 size_tab[HDR], map.size));
268 gst_buffer_unmap (buf, &map);
269 gst_buffer_unref (buf);
274 static GstStateChangeReturn
275 gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
277 GstStateChangeReturn ret;
280 pay = GST_RTP_G723_PAY (element);
282 switch (transition) {
283 case GST_STATE_CHANGE_READY_TO_PAUSED:
284 gst_adapter_clear (pay->adapter);
285 pay->timestamp = GST_CLOCK_TIME_NONE;
293 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
295 switch (transition) {
296 case GST_STATE_CHANGE_PAUSED_TO_READY:
297 gst_adapter_clear (pay->adapter);