2 * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
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.
21 * SECTION:element-rtpac3pay
22 * @see_also: rtpac3depay
24 * Payload AC3 audio into RTP packets according to RFC 4184.
25 * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
28 * <title>Example pipeline</title>
30 * gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
31 * ]| This example pipeline will encode and payload AC3 stream. Refer to
32 * the rtpac3depay example to depayload and decode the RTP stream.
42 #include <gst/rtp/gstrtpbuffer.h>
44 #include "gstrtpac3pay.h"
46 GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug);
47 #define GST_CAT_DEFAULT (rtpac3pay_debug)
49 static GstStaticPadTemplate gst_rtp_ac3_pay_sink_template =
50 GST_STATIC_PAD_TEMPLATE ("sink",
53 GST_STATIC_CAPS ("audio/ac3; " "audio/x-ac3; ")
56 static GstStaticPadTemplate gst_rtp_ac3_pay_src_template =
57 GST_STATIC_PAD_TEMPLATE ("src",
60 GST_STATIC_CAPS ("application/x-rtp, "
61 "media = (string) \"audio\", "
62 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
63 "clock-rate = (int) { 32000, 44100, 48000 }, "
64 "encoding-name = (string) \"AC3\"")
67 static void gst_rtp_ac3_pay_finalize (GObject * object);
69 static GstStateChangeReturn gst_rtp_ac3_pay_change_state (GstElement * element,
70 GstStateChange transition);
72 static gboolean gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload,
74 static gboolean gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload,
76 static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay);
77 static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * payload,
80 #define gst_rtp_ac3_pay_parent_class parent_class
81 G_DEFINE_TYPE (GstRtpAC3Pay, gst_rtp_ac3_pay, GST_TYPE_RTP_BASE_PAYLOAD);
84 gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass)
86 GObjectClass *gobject_class;
87 GstElementClass *gstelement_class;
88 GstRTPBasePayloadClass *gstrtpbasepayload_class;
90 GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0,
91 "AC3 Audio RTP Depayloader");
93 gobject_class = (GObjectClass *) klass;
94 gstelement_class = (GstElementClass *) klass;
95 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
97 gobject_class->finalize = gst_rtp_ac3_pay_finalize;
99 gstelement_class->change_state = gst_rtp_ac3_pay_change_state;
101 gst_element_class_add_pad_template (gstelement_class,
102 gst_static_pad_template_get (&gst_rtp_ac3_pay_src_template));
103 gst_element_class_add_pad_template (gstelement_class,
104 gst_static_pad_template_get (&gst_rtp_ac3_pay_sink_template));
106 gst_element_class_set_static_metadata (gstelement_class,
107 "RTP AC3 audio payloader", "Codec/Payloader/Network/RTP",
108 "Payload AC3 audio as RTP packets (RFC 4184)",
109 "Wim Taymans <wim.taymans@gmail.com>");
111 gstrtpbasepayload_class->set_caps = gst_rtp_ac3_pay_setcaps;
112 gstrtpbasepayload_class->sink_event = gst_rtp_ac3_pay_sink_event;
113 gstrtpbasepayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer;
117 gst_rtp_ac3_pay_init (GstRtpAC3Pay * rtpac3pay)
119 rtpac3pay->adapter = gst_adapter_new ();
123 gst_rtp_ac3_pay_finalize (GObject * object)
125 GstRtpAC3Pay *rtpac3pay;
127 rtpac3pay = GST_RTP_AC3_PAY (object);
129 g_object_unref (rtpac3pay->adapter);
131 G_OBJECT_CLASS (parent_class)->finalize (object);
135 gst_rtp_ac3_pay_reset (GstRtpAC3Pay * pay)
139 gst_adapter_clear (pay->adapter);
140 GST_DEBUG_OBJECT (pay, "reset depayloader");
144 gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
148 GstStructure *structure;
150 structure = gst_caps_get_structure (caps, 0);
152 if (!gst_structure_get_int (structure, "rate", &rate))
153 rate = 90000; /* default */
155 gst_rtp_base_payload_set_options (payload, "audio", TRUE, "AC3", rate);
156 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
162 gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
165 GstRtpAC3Pay *rtpac3pay;
167 rtpac3pay = GST_RTP_AC3_PAY (payload);
169 switch (GST_EVENT_TYPE (event)) {
171 /* make sure we push the last packets in the adapter on EOS */
172 gst_rtp_ac3_pay_flush (rtpac3pay);
174 case GST_EVENT_FLUSH_STOP:
175 gst_rtp_ac3_pay_reset (rtpac3pay);
181 res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
192 static const struct frmsize_s frmsizecod_tbl[] = {
197 {48, {96, 104, 144}},
198 {48, {96, 105, 144}},
199 {56, {112, 121, 168}},
200 {56, {112, 122, 168}},
201 {64, {128, 139, 192}},
202 {64, {128, 140, 192}},
203 {80, {160, 174, 240}},
204 {80, {160, 175, 240}},
205 {96, {192, 208, 288}},
206 {96, {192, 209, 288}},
207 {112, {224, 243, 336}},
208 {112, {224, 244, 336}},
209 {128, {256, 278, 384}},
210 {128, {256, 279, 384}},
211 {160, {320, 348, 480}},
212 {160, {320, 349, 480}},
213 {192, {384, 417, 576}},
214 {192, {384, 418, 576}},
215 {224, {448, 487, 672}},
216 {224, {448, 488, 672}},
217 {256, {512, 557, 768}},
218 {256, {512, 558, 768}},
219 {320, {640, 696, 960}},
220 {320, {640, 697, 960}},
221 {384, {768, 835, 1152}},
222 {384, {768, 836, 1152}},
223 {448, {896, 975, 1344}},
224 {448, {896, 976, 1344}},
225 {512, {1024, 1114, 1536}},
226 {512, {1024, 1115, 1536}},
227 {576, {1152, 1253, 1728}},
228 {576, {1152, 1254, 1728}},
229 {640, {1280, 1393, 1920}},
230 {640, {1280, 1394, 1920}}
234 gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
236 guint avail, FT, NF, mtu;
240 /* the data available in the adapter is either smaller
241 * than the MTU or bigger. In the case it is smaller, the complete
242 * adapter contents can be put in one packet. In the case the
243 * adapter has more than one MTU, we need to split the AC3 data
244 * over multiple packets. */
245 avail = gst_adapter_available (rtpac3pay->adapter);
250 /* number of frames */
253 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpac3pay);
255 GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
262 GstRTPBuffer rtp = { NULL, };
263 GstBuffer *payload_buffer;
265 /* this will be the total length of the packet */
266 packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0);
268 /* fill one MTU or all available bytes */
269 towrite = MIN (packet_len, mtu);
271 /* this is the payload length */
272 payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
274 /* create buffer to hold the payload */
275 outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
278 /* check if it all fits */
279 if (towrite < packet_len) {
282 GST_LOG_OBJECT (rtpac3pay, "we need to fragment");
283 /* check if we will be able to put at least 5/8th of the total
284 * frame in this first frame. */
285 if ((avail * 5) / 8 >= (payload_len - 2))
289 /* check how many fragments we will need */
290 maxlen = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
291 NF = (avail + maxlen - 1) / maxlen;
293 } else if (FT != 3) {
294 /* remaining fragment */
300 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
301 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
303 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
305 * FT: 0: one or more complete frames
306 * 1: initial 5/8 fragment
307 * 2: initial fragment not 5/8
309 * NF: amount of frames if FT = 0, else number of fragments.
311 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
312 GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF);
313 payload = gst_rtp_buffer_get_payload (&rtp);
314 payload[0] = (FT & 3);
318 if (avail == payload_len)
319 gst_rtp_buffer_set_marker (&rtp, TRUE);
320 gst_rtp_buffer_unmap (&rtp);
323 gst_adapter_take_buffer_fast (rtpac3pay->adapter, payload_len);
324 outbuf = gst_buffer_append (outbuf, payload_buffer);
326 avail -= payload_len;
328 GST_BUFFER_PTS (outbuf) = rtpac3pay->first_ts;
329 GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
331 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpac3pay), outbuf);
338 gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * basepayload,
341 GstRtpAC3Pay *rtpac3pay;
343 gsize avail, left, NF;
347 GstClockTime duration, timestamp;
349 rtpac3pay = GST_RTP_AC3_PAY (basepayload);
351 gst_buffer_map (buffer, &map, GST_MAP_READ);
352 duration = GST_BUFFER_DURATION (buffer);
353 timestamp = GST_BUFFER_PTS (buffer);
355 if (GST_BUFFER_IS_DISCONT (buffer)) {
356 GST_DEBUG_OBJECT (rtpac3pay, "DISCONT");
357 gst_rtp_ac3_pay_reset (rtpac3pay);
360 /* count the amount of incomming packets */
365 guint bsid, fscod, frmsizecod, frame_size;
370 if (p[0] != 0x0b || p[1] != 0x77)
377 frmsizecod = p[4] & 0x3f;
380 GST_DEBUG_OBJECT (rtpac3pay, "fscod %u, %u", fscod, frmsizecod);
382 if (fscod >= 3 || frmsizecod >= 38)
385 frame_size = frmsizecod_tbl[frmsizecod].frm_size[fscod] * 2;
386 if (frame_size > left)
390 GST_DEBUG_OBJECT (rtpac3pay, "found frame %" G_GSIZE_FORMAT " of size %u",
396 gst_buffer_unmap (buffer, &map);
400 avail = gst_adapter_available (rtpac3pay->adapter);
402 /* get packet length of previous data and this new data,
403 * payload length includes a 4 byte header */
404 packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + map.size, 0, 0);
406 /* if this buffer is going to overflow the packet, flush what we
408 if (gst_rtp_base_payload_is_filled (basepayload,
409 packet_len, rtpac3pay->duration + duration)) {
410 ret = gst_rtp_ac3_pay_flush (rtpac3pay);
417 GST_DEBUG_OBJECT (rtpac3pay,
418 "first packet, save timestamp %" GST_TIME_FORMAT,
419 GST_TIME_ARGS (timestamp));
420 rtpac3pay->first_ts = timestamp;
421 rtpac3pay->duration = 0;
425 gst_adapter_push (rtpac3pay->adapter, buffer);
426 rtpac3pay->duration += duration;
434 GST_WARNING_OBJECT (rtpac3pay, "no valid AC3 frames found");
439 static GstStateChangeReturn
440 gst_rtp_ac3_pay_change_state (GstElement * element, GstStateChange transition)
442 GstRtpAC3Pay *rtpac3pay;
443 GstStateChangeReturn ret;
445 rtpac3pay = GST_RTP_AC3_PAY (element);
447 switch (transition) {
448 case GST_STATE_CHANGE_READY_TO_PAUSED:
449 gst_rtp_ac3_pay_reset (rtpac3pay);
455 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
457 switch (transition) {
458 case GST_STATE_CHANGE_PAUSED_TO_READY:
459 gst_rtp_ac3_pay_reset (rtpac3pay);
468 gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin)
470 return gst_element_register (plugin, "rtpac3pay",
471 GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_PAY);