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