1 /* GStreamer
2 * Copyright (C) <2006> Wim Taymans <wim.taymans@gmail.com>
3 *
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.
8 *
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.
13 *
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.
18 */
19
20 #ifdef HAVE_CONFIG_H
21 # include "config.h"
22 #endif
23
24 #include <string.h>
25
26 #include <gst/base/gstbitreader.h>
27 #include <gst/rtp/gstrtpbuffer.h>
28
29 #include "gstrtpmp4gpay.h"
30 #include "gstrtputils.h"
31
32 GST_DEBUG_CATEGORY_STATIC (rtpmp4gpay_debug);
33 #define GST_CAT_DEFAULT (rtpmp4gpay_debug)
34
35 static GstStaticPadTemplate gst_rtp_mp4g_pay_sink_template =
36 GST_STATIC_PAD_TEMPLATE ("sink",
37 GST_PAD_SINK,
38 GST_PAD_ALWAYS,
39 GST_STATIC_CAPS ("video/mpeg,"
40 "mpegversion=(int) 4,"
41 "systemstream=(boolean)false;"
42 "audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string) raw")
43 );
44
45 static GstStaticPadTemplate gst_rtp_mp4g_pay_src_template =
46 GST_STATIC_PAD_TEMPLATE ("src",
47 GST_PAD_SRC,
48 GST_PAD_ALWAYS,
49 GST_STATIC_CAPS ("application/x-rtp, "
50 "media = (string) { \"video\", \"audio\", \"application\" }, "
51 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
52 "clock-rate = (int) [1, MAX ], "
53 "encoding-name = (string) \"MPEG4-GENERIC\", "
54 /* required string params */
55 "streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
56 /* "profile-level-id = (string) [1,MAX], " */
57 /* "config = (string) [1,MAX]" */
58 "mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" } "
59 /* Optional general parameters */
60 /* "objecttype = (string) [1,MAX], " */
61 /* "constantsize = (string) [1,MAX], " *//* constant size of each AU */
62 /* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */
63 /* "maxdisplacement = (string) [1,MAX], " */
64 /* "de-interleavebuffersize = (string) [1,MAX], " */
65 /* Optional configuration parameters */
66 /* "sizelength = (string) [1, 16], " *//* max 16 bits, should be enough... */
67 /* "indexlength = (string) [1, 8], " */
68 /* "indexdeltalength = (string) [1, 8], " */
69 /* "ctsdeltalength = (string) [1, 64], " */
70 /* "dtsdeltalength = (string) [1, 64], " */
71 /* "randomaccessindication = (string) {0, 1}, " */
72 /* "streamstateindication = (string) [0, 64], " */
73 /* "auxiliarydatasizelength = (string) [0, 64]" */ )
74 );
75
76
77 static void gst_rtp_mp4g_pay_finalize (GObject * object);
78
79 static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element,
80 GstStateChange transition);
81
82 static gboolean gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload,
83 GstCaps * caps);
84 static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload *
85 payload, GstBuffer * buffer);
86 static gboolean gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload,
87 GstEvent * event);
88
89 #define gst_rtp_mp4g_pay_parent_class parent_class
90 G_DEFINE_TYPE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GST_TYPE_RTP_BASE_PAYLOAD);
91
92 static void
gst_rtp_mp4g_pay_class_init(GstRtpMP4GPayClass * klass)93 gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass)
94 {
95 GObjectClass *gobject_class;
96 GstElementClass *gstelement_class;
97 GstRTPBasePayloadClass *gstrtpbasepayload_class;
98
99 gobject_class = (GObjectClass *) klass;
100 gstelement_class = (GstElementClass *) klass;
101 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
102
103 gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
104
105 gstelement_class->change_state = gst_rtp_mp4g_pay_change_state;
106
107 gstrtpbasepayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
108 gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
109 gstrtpbasepayload_class->sink_event = gst_rtp_mp4g_pay_sink_event;
110
111 gst_element_class_add_static_pad_template (gstelement_class,
112 &gst_rtp_mp4g_pay_src_template);
113 gst_element_class_add_static_pad_template (gstelement_class,
114 &gst_rtp_mp4g_pay_sink_template);
115
116 gst_element_class_set_static_metadata (gstelement_class,
117 "RTP MPEG4 ES payloader",
118 "Codec/Payloader/Network/RTP",
119 "Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
120 "Wim Taymans <wim.taymans@gmail.com>");
121
122 GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0,
123 "MP4-generic RTP Payloader");
124 }
125
126 static void
gst_rtp_mp4g_pay_init(GstRtpMP4GPay * rtpmp4gpay)127 gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay)
128 {
129 rtpmp4gpay->adapter = gst_adapter_new ();
130 }
131
132 static void
gst_rtp_mp4g_pay_reset(GstRtpMP4GPay * rtpmp4gpay)133 gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay)
134 {
135 GST_DEBUG_OBJECT (rtpmp4gpay, "reset");
136
137 gst_adapter_clear (rtpmp4gpay->adapter);
138 }
139
140 static void
gst_rtp_mp4g_pay_cleanup(GstRtpMP4GPay * rtpmp4gpay)141 gst_rtp_mp4g_pay_cleanup (GstRtpMP4GPay * rtpmp4gpay)
142 {
143 gst_rtp_mp4g_pay_reset (rtpmp4gpay);
144
145 g_free (rtpmp4gpay->params);
146 rtpmp4gpay->params = NULL;
147
148 if (rtpmp4gpay->config)
149 gst_buffer_unref (rtpmp4gpay->config);
150 rtpmp4gpay->config = NULL;
151
152 g_free (rtpmp4gpay->profile);
153 rtpmp4gpay->profile = NULL;
154
155 rtpmp4gpay->streamtype = NULL;
156 rtpmp4gpay->mode = NULL;
157
158 rtpmp4gpay->frame_len = 0;
159 }
160
161 static void
gst_rtp_mp4g_pay_finalize(GObject * object)162 gst_rtp_mp4g_pay_finalize (GObject * object)
163 {
164 GstRtpMP4GPay *rtpmp4gpay;
165
166 rtpmp4gpay = GST_RTP_MP4G_PAY (object);
167
168 gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
169
170 g_object_unref (rtpmp4gpay->adapter);
171 rtpmp4gpay->adapter = NULL;
172
173 G_OBJECT_CLASS (parent_class)->finalize (object);
174 }
175
176 static const unsigned int sampling_table[16] = {
177 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
178 16000, 12000, 11025, 8000, 7350, 0, 0, 0
179 };
180
181 static gboolean
gst_rtp_mp4g_pay_parse_audio_config(GstRtpMP4GPay * rtpmp4gpay,GstBuffer * buffer)182 gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
183 GstBuffer * buffer)
184 {
185 GstMapInfo map;
186 guint8 objectType = 0;
187 guint8 samplingIdx = 0;
188 guint8 channelCfg = 0;
189 GstBitReader br;
190
191 gst_buffer_map (buffer, &map, GST_MAP_READ);
192
193 gst_bit_reader_init (&br, map.data, map.size);
194
195 /* any object type is fine, we need to copy it to the profile-level-id field. */
196 if (!gst_bit_reader_get_bits_uint8 (&br, &objectType, 5))
197 goto too_short;
198 if (objectType == 0)
199 goto invalid_object;
200
201 if (!gst_bit_reader_get_bits_uint8 (&br, &samplingIdx, 4))
202 goto too_short;
203 /* only fixed values for now */
204 if (samplingIdx > 12 && samplingIdx != 15)
205 goto wrong_freq;
206
207 if (!gst_bit_reader_get_bits_uint8 (&br, &channelCfg, 4))
208 goto too_short;
209 if (channelCfg > 7)
210 goto wrong_channels;
211
212 /* rtp rate depends on sampling rate of the audio */
213 if (samplingIdx == 15) {
214 guint32 rate = 0;
215
216 /* index of 15 means we get the rate in the next 24 bits */
217 if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
218 goto too_short;
219
220 rtpmp4gpay->rate = rate;
221 } else {
222 /* else use the rate from the table */
223 rtpmp4gpay->rate = sampling_table[samplingIdx];
224 }
225
226 rtpmp4gpay->frame_len = 1024;
227
228 switch (objectType) {
229 case 1:
230 case 2:
231 case 3:
232 case 4:
233 case 6:
234 case 7:
235 {
236 guint8 frameLenFlag = 0;
237
238 if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
239 if (frameLenFlag)
240 rtpmp4gpay->frame_len = 960;
241
242 break;
243 }
244 default:
245 break;
246 }
247
248 /* extra rtp params contain the number of channels */
249 g_free (rtpmp4gpay->params);
250 rtpmp4gpay->params = g_strdup_printf ("%d", channelCfg);
251 /* audio stream type */
252 rtpmp4gpay->streamtype = "5";
253 /* mode only high bitrate for now */
254 rtpmp4gpay->mode = "AAC-hbr";
255 /* profile */
256 g_free (rtpmp4gpay->profile);
257 rtpmp4gpay->profile = g_strdup_printf ("%d", objectType);
258
259 GST_DEBUG_OBJECT (rtpmp4gpay,
260 "objectType: %d, samplingIdx: %d (%d), channelCfg: %d, frame_len %d",
261 objectType, samplingIdx, rtpmp4gpay->rate, channelCfg,
262 rtpmp4gpay->frame_len);
263
264 gst_buffer_unmap (buffer, &map);
265 return TRUE;
266
267 /* ERROR */
268 too_short:
269 {
270 GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
271 (NULL), ("config string too short"));
272 gst_buffer_unmap (buffer, &map);
273 return FALSE;
274 }
275 invalid_object:
276 {
277 GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
278 (NULL), ("invalid object type"));
279 gst_buffer_unmap (buffer, &map);
280 return FALSE;
281 }
282 wrong_freq:
283 {
284 GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
285 (NULL), ("unsupported frequency index %d", samplingIdx));
286 gst_buffer_unmap (buffer, &map);
287 return FALSE;
288 }
289 wrong_channels:
290 {
291 GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
292 (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
293 gst_buffer_unmap (buffer, &map);
294 return FALSE;
295 }
296 }
297
298 #define VOS_STARTCODE 0x000001B0
299
300 static gboolean
gst_rtp_mp4g_pay_parse_video_config(GstRtpMP4GPay * rtpmp4gpay,GstBuffer * buffer)301 gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
302 GstBuffer * buffer)
303 {
304 GstMapInfo map;
305 guint32 code;
306
307 gst_buffer_map (buffer, &map, GST_MAP_READ);
308
309 if (map.size < 5)
310 goto too_short;
311
312 code = GST_READ_UINT32_BE (map.data);
313
314 g_free (rtpmp4gpay->profile);
315 if (code == VOS_STARTCODE) {
316 /* get profile */
317 rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) map.data[4]);
318 } else {
319 GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
320 (NULL), ("profile not found in config string, assuming \'1\'"));
321 rtpmp4gpay->profile = g_strdup ("1");
322 }
323
324 /* fixed rate */
325 rtpmp4gpay->rate = 90000;
326 /* video stream type */
327 rtpmp4gpay->streamtype = "4";
328 /* no params for video */
329 rtpmp4gpay->params = NULL;
330 /* mode */
331 rtpmp4gpay->mode = "generic";
332
333 GST_LOG_OBJECT (rtpmp4gpay, "profile %s", rtpmp4gpay->profile);
334
335 gst_buffer_unmap (buffer, &map);
336
337 return TRUE;
338
339 /* ERROR */
340 too_short:
341 {
342 GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
343 (NULL), ("config string too short"));
344 gst_buffer_unmap (buffer, &map);
345 return FALSE;
346 }
347 }
348
349 static gboolean
gst_rtp_mp4g_pay_new_caps(GstRtpMP4GPay * rtpmp4gpay)350 gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
351 {
352 gchar *config;
353 GValue v = { 0 };
354 gboolean res;
355
356 #define MP4GCAPS \
357 "streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \
358 "profile-level-id", G_TYPE_STRING, rtpmp4gpay->profile, \
359 "mode", G_TYPE_STRING, rtpmp4gpay->mode, \
360 "config", G_TYPE_STRING, config, \
361 "sizelength", G_TYPE_STRING, "13", \
362 "indexlength", G_TYPE_STRING, "3", \
363 "indexdeltalength", G_TYPE_STRING, "3", \
364 NULL
365
366 g_value_init (&v, GST_TYPE_BUFFER);
367 gst_value_set_buffer (&v, rtpmp4gpay->config);
368 config = gst_value_serialize (&v);
369
370 /* hmm, silly */
371 if (rtpmp4gpay->params) {
372 res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
373 "encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
374 } else {
375 res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
376 MP4GCAPS);
377 }
378
379 g_value_unset (&v);
380 g_free (config);
381
382 #undef MP4GCAPS
383 return res;
384 }
385
386 static gboolean
gst_rtp_mp4g_pay_setcaps(GstRTPBasePayload * payload,GstCaps * caps)387 gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
388 {
389 GstRtpMP4GPay *rtpmp4gpay;
390 GstStructure *structure;
391 const GValue *codec_data;
392 const gchar *media_type = NULL;
393 gboolean res;
394
395 rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
396
397 structure = gst_caps_get_structure (caps, 0);
398
399 codec_data = gst_structure_get_value (structure, "codec_data");
400 if (codec_data) {
401 GST_LOG_OBJECT (rtpmp4gpay, "got codec_data");
402 if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
403 GstBuffer *buffer;
404 const gchar *name;
405
406 buffer = gst_value_get_buffer (codec_data);
407 GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data");
408
409 name = gst_structure_get_name (structure);
410
411 /* parse buffer */
412 if (!strcmp (name, "audio/mpeg")) {
413 res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer);
414 media_type = "audio";
415 } else if (!strcmp (name, "video/mpeg")) {
416 res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
417 media_type = "video";
418 } else {
419 res = FALSE;
420 }
421 if (!res)
422 goto config_failed;
423
424 /* now we can configure the buffer */
425 if (rtpmp4gpay->config)
426 gst_buffer_unref (rtpmp4gpay->config);
427
428 rtpmp4gpay->config = gst_buffer_copy (buffer);
429 }
430 }
431 if (media_type == NULL)
432 goto config_failed;
433
434 gst_rtp_base_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
435 rtpmp4gpay->rate);
436
437 res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
438
439 return res;
440
441 /* ERRORS */
442 config_failed:
443 {
444 GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config");
445 return FALSE;
446 }
447 }
448
449 static GstFlowReturn
gst_rtp_mp4g_pay_flush(GstRtpMP4GPay * rtpmp4gpay)450 gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
451 {
452 guint avail, total;
453 GstBuffer *outbuf;
454 GstFlowReturn ret;
455 guint mtu;
456
457 /* the data available in the adapter is either smaller
458 * than the MTU or bigger. In the case it is smaller, the complete
459 * adapter contents can be put in one packet. In the case the
460 * adapter has more than one MTU, we need to fragment the MPEG data
461 * over multiple packets. */
462 total = avail = gst_adapter_available (rtpmp4gpay->adapter);
463
464 ret = GST_FLOW_OK;
465 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4gpay);
466
467 while (avail > 0) {
468 guint towrite;
469 guint8 *payload;
470 guint payload_len;
471 guint packet_len;
472 GstRTPBuffer rtp = { NULL };
473 GstBuffer *paybuf;
474
475 /* this will be the total lenght of the packet */
476 packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
477
478 /* fill one MTU or all available bytes, we need 4 spare bytes for
479 * the AU header. */
480 towrite = MIN (packet_len, mtu - 4);
481
482 /* this is the payload length */
483 payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
484
485 GST_DEBUG_OBJECT (rtpmp4gpay,
486 "avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite,
487 packet_len, payload_len);
488
489 /* create buffer to hold the payload, also make room for the 4 header bytes. */
490 outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
491
492 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
493
494 /* copy payload */
495 payload = gst_rtp_buffer_get_payload (&rtp);
496
497 /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
498 * |AU-headers-length|AU-header|AU-header| |AU-header|padding|
499 * | | (1) | (2) | | (n) | bits |
500 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
501 */
502 /* AU-headers-length, we only have 1 AU-header */
503 payload[0] = 0x00;
504 payload[1] = 0x10; /* we use 16 bits for the header */
505
506 /* +---------------------------------------+
507 * | AU-size |
508 * +---------------------------------------+
509 * | AU-Index / AU-Index-delta |
510 * +---------------------------------------+
511 * | CTS-flag |
512 * +---------------------------------------+
513 * | CTS-delta |
514 * +---------------------------------------+
515 * | DTS-flag |
516 * +---------------------------------------+
517 * | DTS-delta |
518 * +---------------------------------------+
519 * | RAP-flag |
520 * +---------------------------------------+
521 * | Stream-state |
522 * +---------------------------------------+
523 */
524 /* The AU-header, no CTS, DTS, RAP, Stream-state
525 *
526 * AU-size is always the total size of the AU, not the fragmented size
527 */
528 payload[2] = (total & 0x1fe0) >> 5;
529 payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */
530
531 /* marker only if the packet is complete */
532 gst_rtp_buffer_set_marker (&rtp, avail <= payload_len);
533
534 gst_rtp_buffer_unmap (&rtp);
535
536 paybuf = gst_adapter_take_buffer_fast (rtpmp4gpay->adapter, payload_len);
537 gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp4gpay), outbuf, paybuf, 0);
538 outbuf = gst_buffer_append (outbuf, paybuf);
539
540 GST_BUFFER_PTS (outbuf) = rtpmp4gpay->first_timestamp;
541 GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
542
543 GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
544
545 if (rtpmp4gpay->discont) {
546 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
547 /* Only the first outputted buffer has the DISCONT flag */
548 rtpmp4gpay->discont = FALSE;
549 }
550
551 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), outbuf);
552
553 avail -= payload_len;
554 }
555
556 return ret;
557 }
558
559 /* we expect buffers as exactly one complete AU
560 */
561 static GstFlowReturn
gst_rtp_mp4g_pay_handle_buffer(GstRTPBasePayload * basepayload,GstBuffer * buffer)562 gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload * basepayload,
563 GstBuffer * buffer)
564 {
565 GstRtpMP4GPay *rtpmp4gpay;
566
567 rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
568
569 rtpmp4gpay->first_timestamp = GST_BUFFER_PTS (buffer);
570 rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer);
571 rtpmp4gpay->discont = GST_BUFFER_IS_DISCONT (buffer);
572
573 /* we always encode and flush a full AU */
574 gst_adapter_push (rtpmp4gpay->adapter, buffer);
575
576 return gst_rtp_mp4g_pay_flush (rtpmp4gpay);
577 }
578
579 static gboolean
gst_rtp_mp4g_pay_sink_event(GstRTPBasePayload * payload,GstEvent * event)580 gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
581 {
582 GstRtpMP4GPay *rtpmp4gpay;
583
584 rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
585
586 GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
587
588 switch (GST_EVENT_TYPE (event)) {
589 case GST_EVENT_SEGMENT:
590 case GST_EVENT_EOS:
591 /* This flush call makes sure that the last buffer is always pushed
592 * to the base payloader */
593 gst_rtp_mp4g_pay_flush (rtpmp4gpay);
594 break;
595 case GST_EVENT_FLUSH_STOP:
596 gst_rtp_mp4g_pay_reset (rtpmp4gpay);
597 break;
598 default:
599 break;
600 }
601
602 /* let parent handle event too */
603 return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
604 }
605
606 static GstStateChangeReturn
gst_rtp_mp4g_pay_change_state(GstElement * element,GstStateChange transition)607 gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition)
608 {
609 GstStateChangeReturn ret;
610 GstRtpMP4GPay *rtpmp4gpay;
611
612 rtpmp4gpay = GST_RTP_MP4G_PAY (element);
613
614 switch (transition) {
615 case GST_STATE_CHANGE_READY_TO_PAUSED:
616 gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
617 break;
618 default:
619 break;
620 }
621
622 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
623
624 switch (transition) {
625 case GST_STATE_CHANGE_PAUSED_TO_READY:
626 gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
627 break;
628 default:
629 break;
630 }
631
632 return ret;
633 }
634
635 gboolean
gst_rtp_mp4g_pay_plugin_init(GstPlugin * plugin)636 gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
637 {
638 return gst_element_register (plugin, "rtpmp4gpay",
639 GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_PAY);
640 }
641