2 * Copyright (C) 2008 Wim Taymans <wim.taymans at 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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
23 #include <gst/app/gstappsrc.h>
24 #include <gst/app/gstappsink.h>
26 #include "rtsp-media.h"
28 #define DEFAULT_SHARED FALSE
29 #define DEFAULT_REUSABLE FALSE
30 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
31 //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
32 #define DEFAULT_EOS_SHUTDOWN FALSE
33 #define DEFAULT_BUFFER_SIZE 0x80000
34 #define DEFAULT_MULTICAST_GROUP "224.2.0.1"
36 /* define to dump received RTCP packets */
59 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
60 #define GST_CAT_DEFAULT rtsp_media_debug
62 static GQuark ssrc_stream_map_key;
64 static void gst_rtsp_media_get_property (GObject * object, guint propid,
65 GValue * value, GParamSpec * pspec);
66 static void gst_rtsp_media_set_property (GObject * object, guint propid,
67 const GValue * value, GParamSpec * pspec);
68 static void gst_rtsp_media_finalize (GObject * obj);
70 static gpointer do_loop (GstRTSPMediaClass * klass);
71 static gboolean default_handle_message (GstRTSPMedia * media,
72 GstMessage * message);
73 static gboolean default_unprepare (GstRTSPMedia * media);
74 static void unlock_streams (GstRTSPMedia * media);
75 static void default_handle_mtu (GstRTSPMedia * media, guint mtu);
77 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
79 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
82 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
84 GObjectClass *gobject_class;
86 gobject_class = G_OBJECT_CLASS (klass);
88 gobject_class->get_property = gst_rtsp_media_get_property;
89 gobject_class->set_property = gst_rtsp_media_set_property;
90 gobject_class->finalize = gst_rtsp_media_finalize;
92 g_object_class_install_property (gobject_class, PROP_SHARED,
93 g_param_spec_boolean ("shared", "Shared",
94 "If this media pipeline can be shared", DEFAULT_SHARED,
95 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
97 g_object_class_install_property (gobject_class, PROP_REUSABLE,
98 g_param_spec_boolean ("reusable", "Reusable",
99 "If this media pipeline can be reused after an unprepare",
100 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
102 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
103 g_param_spec_flags ("protocols", "Protocols",
104 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
105 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
107 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
108 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
109 "Send an EOS event to the pipeline before unpreparing",
110 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
112 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
113 g_param_spec_uint ("buffer-size", "Buffer Size",
114 "The kernel UDP buffer size to use", 0, G_MAXUINT,
115 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
117 g_object_class_install_property (gobject_class, PROP_MULTICAST_GROUP,
118 g_param_spec_string ("multicast-group", "Multicast Group",
119 "The Multicast group to send media to",
120 DEFAULT_MULTICAST_GROUP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
122 gst_rtsp_media_signals[SIGNAL_PREPARED] =
123 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
124 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
125 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
127 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
128 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
129 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
130 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
132 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
133 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
134 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
135 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
137 klass->context = g_main_context_new ();
138 klass->loop = g_main_loop_new (klass->context, TRUE);
140 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
142 klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
144 klass->handle_message = default_handle_message;
145 klass->unprepare = default_unprepare;
146 klass->handle_mtu = default_handle_mtu;
148 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
152 gst_rtsp_media_init (GstRTSPMedia * media)
154 media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
155 g_mutex_init (&media->lock);
156 g_cond_init (&media->cond);
158 media->shared = DEFAULT_SHARED;
159 media->reusable = DEFAULT_REUSABLE;
160 media->protocols = DEFAULT_PROTOCOLS;
161 media->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
162 media->buffer_size = DEFAULT_BUFFER_SIZE;
163 media->multicast_group = g_strdup (DEFAULT_MULTICAST_GROUP);
167 gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans * trans)
169 if (trans->transport) {
170 gst_rtsp_transport_free (trans->transport);
171 trans->transport = NULL;
173 if (trans->rtpsource) {
174 g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
175 trans->rtpsource = NULL;
180 gst_rtsp_media_stream_free (GstRTSPMediaStream * stream)
183 g_object_unref (stream->session);
186 gst_caps_unref (stream->caps);
188 if (stream->send_rtp_sink)
189 gst_object_unref (stream->send_rtp_sink);
190 if (stream->send_rtp_src)
191 gst_object_unref (stream->send_rtp_src);
192 if (stream->send_rtcp_src)
193 gst_object_unref (stream->send_rtcp_src);
194 if (stream->recv_rtcp_sink)
195 gst_object_unref (stream->recv_rtcp_sink);
196 if (stream->recv_rtp_sink)
197 gst_object_unref (stream->recv_rtp_sink);
199 g_list_free (stream->transports);
205 gst_rtsp_media_finalize (GObject * obj)
210 media = GST_RTSP_MEDIA (obj);
212 GST_INFO ("finalize media %p", media);
214 if (media->pipeline) {
215 unlock_streams (media);
216 gst_element_set_state (media->pipeline, GST_STATE_NULL);
217 gst_object_unref (media->pipeline);
220 for (i = 0; i < media->streams->len; i++) {
221 GstRTSPMediaStream *stream;
223 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
225 gst_rtsp_media_stream_free (stream);
227 g_array_free (media->streams, TRUE);
229 g_list_foreach (media->dynamic, (GFunc) gst_object_unref, NULL);
230 g_list_free (media->dynamic);
233 g_source_destroy (media->source);
234 g_source_unref (media->source);
236 g_free (media->multicast_group);
237 g_mutex_clear (&media->lock);
238 g_cond_clear (&media->cond);
240 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
244 gst_rtsp_media_get_property (GObject * object, guint propid,
245 GValue * value, GParamSpec * pspec)
247 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
251 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
254 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
257 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
259 case PROP_EOS_SHUTDOWN:
260 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
262 case PROP_BUFFER_SIZE:
263 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
265 case PROP_MULTICAST_GROUP:
266 g_value_take_string (value, gst_rtsp_media_get_multicast_group (media));
269 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
274 gst_rtsp_media_set_property (GObject * object, guint propid,
275 const GValue * value, GParamSpec * pspec)
277 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
281 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
284 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
287 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
289 case PROP_EOS_SHUTDOWN:
290 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
292 case PROP_BUFFER_SIZE:
293 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
295 case PROP_MULTICAST_GROUP:
296 gst_rtsp_media_set_multicast_group (media, g_value_get_string (value));
299 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
304 do_loop (GstRTSPMediaClass * klass)
306 GST_INFO ("enter mainloop");
307 g_main_loop_run (klass->loop);
308 GST_INFO ("exit mainloop");
314 collect_media_stats (GstRTSPMedia * media)
316 gint64 position, duration;
318 media->range.unit = GST_RTSP_RANGE_NPT;
320 if (media->is_live) {
321 media->range.min.type = GST_RTSP_TIME_NOW;
322 media->range.min.seconds = -1;
323 media->range.max.type = GST_RTSP_TIME_END;
324 media->range.max.seconds = -1;
326 /* get the position */
327 if (!gst_element_query_position (media->pipeline, GST_FORMAT_TIME,
329 GST_INFO ("position query failed");
333 /* get the duration */
334 if (!gst_element_query_duration (media->pipeline, GST_FORMAT_TIME,
336 GST_INFO ("duration query failed");
340 GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
341 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
343 if (position == -1) {
344 media->range.min.type = GST_RTSP_TIME_NOW;
345 media->range.min.seconds = -1;
347 media->range.min.type = GST_RTSP_TIME_SECONDS;
348 media->range.min.seconds = ((gdouble) position) / GST_SECOND;
350 if (duration == -1) {
351 media->range.max.type = GST_RTSP_TIME_END;
352 media->range.max.seconds = -1;
354 media->range.max.type = GST_RTSP_TIME_SECONDS;
355 media->range.max.seconds = ((gdouble) duration) / GST_SECOND;
361 * gst_rtsp_media_new:
363 * Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
364 * element to produce RTP data for one or more related (audio/video/..)
367 * Returns: a new #GstRTSPMedia object.
370 gst_rtsp_media_new (void)
372 GstRTSPMedia *result;
374 result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
380 * gst_rtsp_media_set_shared:
381 * @media: a #GstRTSPMedia
382 * @shared: the new value
384 * Set or unset if the pipeline for @media can be shared will multiple clients.
385 * When @shared is %TRUE, client requests for this media will share the media
389 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
391 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
393 media->shared = shared;
397 * gst_rtsp_media_is_shared:
398 * @media: a #GstRTSPMedia
400 * Check if the pipeline for @media can be shared between multiple clients.
402 * Returns: %TRUE if the media can be shared between clients.
405 gst_rtsp_media_is_shared (GstRTSPMedia * media)
407 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
409 return media->shared;
413 * gst_rtsp_media_set_reusable:
414 * @media: a #GstRTSPMedia
415 * @reusable: the new value
417 * Set or unset if the pipeline for @media can be reused after the pipeline has
421 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
423 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
425 media->reusable = reusable;
429 * gst_rtsp_media_is_reusable:
430 * @media: a #GstRTSPMedia
432 * Check if the pipeline for @media can be reused after an unprepare.
434 * Returns: %TRUE if the media can be reused
437 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
439 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
441 return media->reusable;
445 * gst_rtsp_media_set_protocols:
446 * @media: a #GstRTSPMedia
447 * @protocols: the new flags
449 * Configure the allowed lower transport for @media.
452 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
454 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
456 media->protocols = protocols;
460 * gst_rtsp_media_get_protocols:
461 * @media: a #GstRTSPMedia
463 * Get the allowed protocols of @media.
465 * Returns: a #GstRTSPLowerTrans
468 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
470 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
471 GST_RTSP_LOWER_TRANS_UNKNOWN);
473 return media->protocols;
477 * gst_rtsp_media_set_eos_shutdown:
478 * @media: a #GstRTSPMedia
479 * @eos_shutdown: the new value
481 * Set or unset if an EOS event will be sent to the pipeline for @media before
485 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
487 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
489 media->eos_shutdown = eos_shutdown;
493 * gst_rtsp_media_is_eos_shutdown:
494 * @media: a #GstRTSPMedia
496 * Check if the pipeline for @media will send an EOS down the pipeline before
499 * Returns: %TRUE if the media will send EOS before unpreparing.
502 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
504 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
506 return media->eos_shutdown;
510 * gst_rtsp_media_set_buffer_size:
511 * @media: a #GstRTSPMedia
512 * @size: the new value
514 * Set the kernel UDP buffer size.
517 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
519 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
521 media->buffer_size = size;
525 * gst_rtsp_media_get_buffer_size:
526 * @media: a #GstRTSPMedia
528 * Get the kernel UDP buffer size.
530 * Returns: the kernel UDP buffer size.
533 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
535 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
537 return media->buffer_size;
541 * gst_rtsp_media_set_multicast_group:
542 * @media: a #GstRTSPMedia
543 * @mc: the new multicast group
545 * Set the multicast group that media from @media will be streamed to.
548 gst_rtsp_media_set_multicast_group (GstRTSPMedia * media, const gchar * mc)
550 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
552 g_mutex_lock (&media->lock);
553 g_free (media->multicast_group);
554 media->multicast_group = g_strdup (mc);
555 g_mutex_unlock (&media->lock);
559 * gst_rtsp_media_get_multicast_group:
560 * @media: a #GstRTSPMedia
562 * Get the multicast group that media from @media will be streamed to.
564 * Returns: the multicast group
567 gst_rtsp_media_get_multicast_group (GstRTSPMedia * media)
571 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
573 g_mutex_lock (&media->lock);
574 result = g_strdup (media->multicast_group);
575 g_mutex_unlock (&media->lock);
581 * gst_rtsp_media_set_auth:
582 * @media: a #GstRTSPMedia
583 * @auth: a #GstRTSPAuth
585 * configure @auth to be used as the authentication manager of @media.
588 gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
592 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
601 g_object_unref (old);
606 * gst_rtsp_media_get_auth:
607 * @media: a #GstRTSPMedia
609 * Get the #GstRTSPAuth used as the authentication manager of @media.
611 * Returns: the #GstRTSPAuth of @media. g_object_unref() after
615 gst_rtsp_media_get_auth (GstRTSPMedia * media)
619 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
621 if ((result = media->auth))
622 g_object_ref (result);
629 * gst_rtsp_media_n_streams:
630 * @media: a #GstRTSPMedia
632 * Get the number of streams in this media.
634 * Returns: The number of streams.
637 gst_rtsp_media_n_streams (GstRTSPMedia * media)
639 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
641 return media->streams->len;
645 * gst_rtsp_media_get_stream:
646 * @media: a #GstRTSPMedia
647 * @idx: the stream index
649 * Retrieve the stream with index @idx from @media.
651 * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
652 * that index did not exist.
655 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
657 GstRTSPMediaStream *res;
659 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
661 if (idx < media->streams->len)
662 res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
670 * gst_rtsp_media_get_range_string:
671 * @media: a #GstRTSPMedia
672 * @play: for the PLAY request
674 * Get the current range as a string.
676 * Returns: The range as a string, g_free() after usage.
679 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play)
682 GstRTSPTimeRange range;
685 range = media->range;
687 if (!play && media->active > 0) {
688 range.min.type = GST_RTSP_TIME_NOW;
689 range.min.seconds = -1;
692 result = gst_rtsp_range_to_string (&range);
698 * gst_rtsp_media_seek:
699 * @media: a #GstRTSPMedia
700 * @range: a #GstRTSPTimeRange
702 * Seek the pipeline to @range.
704 * Returns: %TRUE on success.
707 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
712 GstSeekType start_type, stop_type;
714 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
715 g_return_val_if_fail (range != NULL, FALSE);
717 if (!media->seekable) {
718 GST_INFO ("pipeline is not seekable");
722 if (range->unit != GST_RTSP_RANGE_NPT)
725 /* depends on the current playing state of the pipeline. We might need to
726 * queue this until we get EOS. */
727 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
729 start_type = stop_type = GST_SEEK_TYPE_NONE;
731 switch (range->min.type) {
732 case GST_RTSP_TIME_NOW:
735 case GST_RTSP_TIME_SECONDS:
736 /* only seek when something changed */
737 if (media->range.min.seconds == range->min.seconds) {
740 start = range->min.seconds * GST_SECOND;
741 start_type = GST_SEEK_TYPE_SET;
744 case GST_RTSP_TIME_END:
748 switch (range->max.type) {
749 case GST_RTSP_TIME_SECONDS:
750 /* only seek when something changed */
751 if (media->range.max.seconds == range->max.seconds) {
754 stop = range->max.seconds * GST_SECOND;
755 stop_type = GST_SEEK_TYPE_SET;
758 case GST_RTSP_TIME_END:
760 stop_type = GST_SEEK_TYPE_SET;
762 case GST_RTSP_TIME_NOW:
767 if (start != -1 || stop != -1) {
768 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
769 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
771 res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
772 flags, start_type, start, stop_type, stop);
774 /* and block for the seek to complete */
775 GST_INFO ("done seeking %d", res);
776 gst_element_get_state (media->pipeline, NULL, NULL, -1);
777 GST_INFO ("prerolled again");
779 collect_media_stats (media);
781 GST_INFO ("no seek needed");
790 GST_WARNING ("seek unit %d not supported", range->unit);
795 GST_WARNING ("weird range type %d not supported", range->min.type);
801 * gst_rtsp_media_stream_rtp:
802 * @stream: a #GstRTSPMediaStream
803 * @buffer: a #GstBuffer
805 * Handle an RTP buffer for the stream. This method is usually called when a
806 * message has been received from a client using the TCP transport.
808 * This function takes ownership of @buffer.
810 * Returns: a GstFlowReturn.
813 gst_rtsp_media_stream_rtp (GstRTSPMediaStream * stream, GstBuffer * buffer)
817 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
823 * gst_rtsp_media_stream_rtcp:
824 * @stream: a #GstRTSPMediaStream
825 * @buffer: a #GstBuffer
827 * Handle an RTCP buffer for the stream. This method is usually called when a
828 * message has been received from a client using the TCP transport.
830 * This function takes ownership of @buffer.
832 * Returns: a GstFlowReturn.
835 gst_rtsp_media_stream_rtcp (GstRTSPMediaStream * stream, GstBuffer * buffer)
839 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
844 /* Allocate the udp ports and sockets */
846 alloc_udp_ports (GstRTSPMedia * media, GstRTSPMediaStream * stream)
848 GstStateChangeReturn ret;
849 GstElement *udpsrc0, *udpsrc1;
850 GstElement *udpsink0, *udpsink1;
851 gint tmp_rtp, tmp_rtcp;
853 gint rtpport, rtcpport;
863 /* Start with random port */
867 host = "udp://[::0]";
869 host = "udp://0.0.0.0";
871 /* try to allocate 2 UDP ports, the RTP port should be an even
872 * number and the RTCP port should be the next (uneven) port */
874 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
876 goto no_udp_protocol;
877 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
879 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
880 if (ret == GST_STATE_CHANGE_FAILURE) {
886 gst_element_set_state (udpsrc0, GST_STATE_NULL);
887 gst_object_unref (udpsrc0);
891 goto no_udp_protocol;
894 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
896 /* check if port is even */
897 if ((tmp_rtp & 1) != 0) {
898 /* port not even, close and allocate another */
902 gst_element_set_state (udpsrc0, GST_STATE_NULL);
903 gst_object_unref (udpsrc0);
909 /* allocate port+1 for RTCP now */
910 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
912 goto no_udp_rtcp_protocol;
915 tmp_rtcp = tmp_rtp + 1;
916 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
918 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
919 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
920 if (ret == GST_STATE_CHANGE_FAILURE) {
925 gst_element_set_state (udpsrc0, GST_STATE_NULL);
926 gst_object_unref (udpsrc0);
928 gst_element_set_state (udpsrc1, GST_STATE_NULL);
929 gst_object_unref (udpsrc1);
935 /* all fine, do port check */
936 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
937 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
939 /* this should not happen... */
940 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
943 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
945 goto no_udp_protocol;
947 g_object_get (G_OBJECT (udpsrc0), "socket", &socket, NULL);
948 g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
949 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
951 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
953 goto no_udp_protocol;
955 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
956 "send-duplicates")) {
957 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
958 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
961 ("old multiudpsink version found without send-duplicates property");
964 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
966 g_object_set (G_OBJECT (udpsink0), "buffer-size", media->buffer_size, NULL);
968 GST_WARNING ("multiudpsink version found without buffer-size property");
971 g_object_get (G_OBJECT (udpsrc1), "socket", &socket, NULL);
972 g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
973 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
974 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
975 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
977 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
978 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
979 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
980 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
982 /* we keep these elements, we configure all in configure_transport when the
983 * server told us to really use the UDP ports. */
984 stream->udpsrc[0] = udpsrc0;
985 stream->udpsrc[1] = udpsrc1;
986 stream->udpsink[0] = udpsink0;
987 stream->udpsink[1] = udpsink1;
988 stream->server_port.min = rtpport;
989 stream->server_port.max = rtcpport;
1002 no_udp_rtcp_protocol:
1013 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1014 gst_object_unref (udpsrc0);
1017 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1018 gst_object_unref (udpsrc1);
1021 gst_element_set_state (udpsink0, GST_STATE_NULL);
1022 gst_object_unref (udpsink0);
1025 gst_element_set_state (udpsink1, GST_STATE_NULL);
1026 gst_object_unref (udpsink1);
1032 /* executed from streaming thread */
1034 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
1037 GstCaps *newcaps, *oldcaps;
1039 newcaps = gst_pad_get_current_caps (pad);
1041 oldcaps = stream->caps;
1042 stream->caps = newcaps;
1045 gst_caps_unref (oldcaps);
1047 capsstr = gst_caps_to_string (newcaps);
1048 GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr);
1053 dump_structure (const GstStructure * s)
1057 sstr = gst_structure_to_string (s);
1058 GST_INFO ("structure: %s", sstr);
1062 static GstRTSPMediaTrans *
1063 find_transport (GstRTSPMediaStream * stream, const gchar * rtcp_from)
1066 GstRTSPMediaTrans *result = NULL;
1071 if (rtcp_from == NULL)
1074 tmp = g_strrstr (rtcp_from, ":");
1078 port = atoi (tmp + 1);
1079 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1081 GST_INFO ("finding %s:%d in %d transports", dest, port,
1082 g_list_length (stream->transports));
1084 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
1085 GstRTSPMediaTrans *trans = walk->data;
1088 min = trans->transport->client_port.min;
1089 max = trans->transport->client_port.max;
1091 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
1102 static GstRTSPMediaTrans *
1103 check_transport (GObject * source, GstRTSPMediaStream * stream)
1105 GstStructure *stats;
1106 GstRTSPMediaTrans *trans;
1108 /* see if we have a stream to match with the origin of the RTCP packet */
1109 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1110 if (trans == NULL) {
1111 g_object_get (source, "stats", &stats, NULL);
1113 const gchar *rtcp_from;
1115 dump_structure (stats);
1117 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1118 if ((trans = find_transport (stream, rtcp_from))) {
1119 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1122 /* keep ref to the source */
1123 trans->rtpsource = source;
1125 g_object_set_qdata (source, ssrc_stream_map_key, trans);
1127 gst_structure_free (stats);
1135 on_new_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1137 GstRTSPMediaTrans *trans;
1139 GST_INFO ("%p: new source %p", stream, source);
1141 trans = check_transport (source, stream);
1144 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1148 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1150 GST_INFO ("%p: new SDES %p", stream, source);
1154 on_ssrc_active (GObject * session, GObject * source,
1155 GstRTSPMediaStream * stream)
1157 GstRTSPMediaTrans *trans;
1159 trans = check_transport (source, stream);
1162 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1164 if (trans && trans->keep_alive)
1165 trans->keep_alive (trans->ka_user_data);
1169 GstStructure *stats;
1170 g_object_get (source, "stats", &stats, NULL);
1172 dump_structure (stats);
1173 gst_structure_free (stats);
1180 on_bye_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1182 GST_INFO ("%p: source %p bye", stream, source);
1186 on_bye_timeout (GObject * session, GObject * source,
1187 GstRTSPMediaStream * stream)
1189 GstRTSPMediaTrans *trans;
1191 GST_INFO ("%p: source %p bye timeout", stream, source);
1193 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1194 trans->rtpsource = NULL;
1195 trans->timeout = TRUE;
1200 on_timeout (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1202 GstRTSPMediaTrans *trans;
1204 GST_INFO ("%p: source %p timeout", stream, source);
1206 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1207 trans->rtpsource = NULL;
1208 trans->timeout = TRUE;
1212 static GstFlowReturn
1213 handle_new_sample (GstAppSink * sink, gpointer user_data)
1218 GstRTSPMediaStream *stream;
1220 sample = gst_app_sink_pull_sample (sink);
1224 stream = (GstRTSPMediaStream *) user_data;
1225 buffer = gst_sample_get_buffer (sample);
1227 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
1228 GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
1230 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
1232 tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
1235 tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
1238 gst_sample_unref (sample);
1243 static GstAppSinkCallbacks sink_cb = {
1244 NULL, /* not interested in EOS */
1245 NULL, /* not interested in preroll samples */
1249 /* prepare the pipeline objects to handle @stream in @media */
1251 setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media)
1254 GstPad *pad, *teepad, *queuepad, *selpad;
1255 GstPadLinkReturn ret;
1258 /* allocate udp ports, we will have 4 of them, 2 for receiving RTP/RTCP and 2
1259 * for sending RTP/RTCP. The sender and receiver ports are shared between the
1261 if (!alloc_udp_ports (media, stream))
1264 /* add the ports to the pipeline */
1265 for (i = 0; i < 2; i++) {
1266 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[i]);
1267 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[i]);
1270 /* create elements for the TCP transfer */
1271 for (i = 0; i < 2; i++) {
1272 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1273 stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
1274 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
1275 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1276 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
1277 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appqueue[i]);
1278 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
1279 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
1280 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
1281 &sink_cb, stream, NULL);
1284 /* hook up the stream to the RTP session elements. */
1285 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1286 stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1288 name = g_strdup_printf ("send_rtp_src_%u", idx);
1289 stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
1291 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1292 stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
1294 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1295 stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
1297 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1298 stream->recv_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1301 /* get the session */
1302 g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
1305 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1307 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1309 g_signal_connect (stream->session, "on-ssrc-active",
1310 (GCallback) on_ssrc_active, stream);
1311 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1313 g_signal_connect (stream->session, "on-bye-timeout",
1314 (GCallback) on_bye_timeout, stream);
1315 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
1318 /* link the RTP pad to the session manager */
1319 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
1320 if (ret != GST_PAD_LINK_OK)
1323 /* make tee for RTP and link to stream */
1324 stream->tee[0] = gst_element_factory_make ("tee", NULL);
1325 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[0]);
1327 pad = gst_element_get_static_pad (stream->tee[0], "sink");
1328 gst_pad_link (stream->send_rtp_src, pad);
1329 gst_object_unref (pad);
1331 /* link RTP sink, we're pretty sure this will work. */
1332 teepad = gst_element_get_request_pad (stream->tee[0], "src_%u");
1333 pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
1334 gst_pad_link (teepad, pad);
1335 gst_object_unref (pad);
1336 gst_object_unref (teepad);
1338 teepad = gst_element_get_request_pad (stream->tee[0], "src_%u");
1339 pad = gst_element_get_static_pad (stream->appqueue[0], "sink");
1340 gst_pad_link (teepad, pad);
1341 gst_object_unref (pad);
1342 gst_object_unref (teepad);
1344 queuepad = gst_element_get_static_pad (stream->appqueue[0], "src");
1345 pad = gst_element_get_static_pad (stream->appsink[0], "sink");
1346 gst_pad_link (queuepad, pad);
1347 gst_object_unref (pad);
1348 gst_object_unref (queuepad);
1350 /* make tee for RTCP */
1351 stream->tee[1] = gst_element_factory_make ("tee", NULL);
1352 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[1]);
1354 pad = gst_element_get_static_pad (stream->tee[1], "sink");
1355 gst_pad_link (stream->send_rtcp_src, pad);
1356 gst_object_unref (pad);
1358 /* link RTCP elements */
1359 teepad = gst_element_get_request_pad (stream->tee[1], "src_%u");
1360 pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
1361 gst_pad_link (teepad, pad);
1362 gst_object_unref (pad);
1363 gst_object_unref (teepad);
1365 teepad = gst_element_get_request_pad (stream->tee[1], "src_%u");
1366 pad = gst_element_get_static_pad (stream->appqueue[1], "sink");
1367 gst_pad_link (teepad, pad);
1368 gst_object_unref (pad);
1369 gst_object_unref (teepad);
1371 queuepad = gst_element_get_static_pad (stream->appqueue[1], "src");
1372 pad = gst_element_get_static_pad (stream->appsink[1], "sink");
1373 gst_pad_link (queuepad, pad);
1374 gst_object_unref (pad);
1375 gst_object_unref (queuepad);
1377 /* make selector for the RTP receivers */
1378 stream->selector[0] = gst_element_factory_make ("funnel", NULL);
1379 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]);
1381 pad = gst_element_get_static_pad (stream->selector[0], "src");
1382 gst_pad_link (pad, stream->recv_rtp_sink);
1383 gst_object_unref (pad);
1385 selpad = gst_element_get_request_pad (stream->selector[0], "sink_%u");
1386 pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
1387 gst_pad_link (pad, selpad);
1388 gst_object_unref (pad);
1389 gst_object_unref (selpad);
1391 selpad = gst_element_get_request_pad (stream->selector[0], "sink_%u");
1392 pad = gst_element_get_static_pad (stream->appsrc[0], "src");
1393 gst_pad_link (pad, selpad);
1394 gst_object_unref (pad);
1395 gst_object_unref (selpad);
1397 /* make selector for the RTCP receivers */
1398 stream->selector[1] = gst_element_factory_make ("funnel", NULL);
1399 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]);
1401 pad = gst_element_get_static_pad (stream->selector[1], "src");
1402 gst_pad_link (pad, stream->recv_rtcp_sink);
1403 gst_object_unref (pad);
1405 selpad = gst_element_get_request_pad (stream->selector[1], "sink_%u");
1406 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
1407 gst_pad_link (pad, selpad);
1408 gst_object_unref (pad);
1409 gst_object_unref (selpad);
1411 selpad = gst_element_get_request_pad (stream->selector[1], "sink_%u");
1412 pad = gst_element_get_static_pad (stream->appsrc[1], "src");
1413 gst_pad_link (pad, selpad);
1414 gst_object_unref (pad);
1415 gst_object_unref (selpad);
1417 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1419 gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
1420 gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
1421 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
1422 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
1424 /* be notified of caps changes */
1425 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
1426 (GCallback) caps_notify, stream);
1428 stream->prepared = TRUE;
1435 GST_WARNING ("failed to link stream %d", idx);
1441 unlock_streams (GstRTSPMedia * media)
1445 /* unlock the udp src elements */
1446 n_streams = gst_rtsp_media_n_streams (media);
1447 for (i = 0; i < n_streams; i++) {
1448 GstRTSPMediaStream *stream;
1450 stream = gst_rtsp_media_get_stream (media, i);
1452 gst_element_set_locked_state (stream->udpsrc[0], FALSE);
1453 gst_element_set_locked_state (stream->udpsrc[1], FALSE);
1458 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1460 g_mutex_lock (&media->lock);
1461 /* never overwrite the error status */
1462 if (media->status != GST_RTSP_MEDIA_STATUS_ERROR)
1463 media->status = status;
1464 GST_DEBUG ("setting new status to %d", status);
1465 g_cond_broadcast (&media->cond);
1466 g_mutex_unlock (&media->lock);
1469 static GstRTSPMediaStatus
1470 gst_rtsp_media_get_status (GstRTSPMedia * media)
1472 GstRTSPMediaStatus result;
1475 g_mutex_lock (&media->lock);
1476 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1477 /* while we are preparing, wait */
1478 while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1479 GST_DEBUG ("waiting for status change");
1480 if (!g_cond_wait_until (&media->cond, &media->lock, end_time)) {
1481 GST_DEBUG ("timeout, assuming error status");
1482 media->status = GST_RTSP_MEDIA_STATUS_ERROR;
1485 /* could be success or error */
1486 result = media->status;
1487 GST_DEBUG ("got status %d", result);
1488 g_mutex_unlock (&media->lock);
1494 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1496 GstMessageType type;
1498 type = GST_MESSAGE_TYPE (message);
1501 case GST_MESSAGE_STATE_CHANGED:
1503 case GST_MESSAGE_BUFFERING:
1507 gst_message_parse_buffering (message, &percent);
1509 /* no state management needed for live pipelines */
1513 if (percent == 100) {
1514 /* a 100% message means buffering is done */
1515 media->buffering = FALSE;
1516 /* if the desired state is playing, go back */
1517 if (media->target_state == GST_STATE_PLAYING) {
1518 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1519 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1521 GST_INFO ("Buffering done");
1524 /* buffering busy */
1525 if (media->buffering == FALSE) {
1526 if (media->target_state == GST_STATE_PLAYING) {
1527 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1528 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1529 gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1531 GST_INFO ("Buffering ...");
1534 media->buffering = TRUE;
1538 case GST_MESSAGE_LATENCY:
1540 gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline));
1543 case GST_MESSAGE_ERROR:
1548 gst_message_parse_error (message, &gerror, &debug);
1549 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1550 g_error_free (gerror);
1553 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1556 case GST_MESSAGE_WARNING:
1561 gst_message_parse_warning (message, &gerror, &debug);
1562 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1563 g_error_free (gerror);
1567 case GST_MESSAGE_ELEMENT:
1569 case GST_MESSAGE_STREAM_STATUS:
1571 case GST_MESSAGE_ASYNC_DONE:
1572 if (!media->adding) {
1573 /* when we are dynamically adding pads, the addition of the udpsrc will
1574 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1575 * wait for the final ASYNC_DONE after everything prerolled */
1576 GST_INFO ("%p: got ASYNC_DONE", media);
1577 collect_media_stats (media);
1579 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1581 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1584 case GST_MESSAGE_EOS:
1585 GST_INFO ("%p: got EOS", media);
1586 if (media->eos_pending) {
1587 GST_DEBUG ("shutting down after EOS");
1588 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1589 media->eos_pending = FALSE;
1590 g_object_unref (media);
1594 GST_INFO ("%p: got message type %s", media,
1595 gst_message_type_get_name (type));
1602 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1604 GstRTSPMediaClass *klass;
1607 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1609 if (klass->handle_message)
1610 ret = klass->handle_message (media, message);
1617 /* called from streaming threads */
1619 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1621 GstRTSPMediaStream *stream;
1625 i = media->streams->len + 1;
1627 GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
1629 stream = g_new0 (GstRTSPMediaStream, 1);
1630 stream->payloader = element;
1632 name = g_strdup_printf ("dynpay%d", i);
1634 media->adding = TRUE;
1636 /* ghost the pad of the payloader to the element */
1637 stream->srcpad = gst_ghost_pad_new (name, pad);
1638 gst_pad_set_active (stream->srcpad, TRUE);
1639 gst_element_add_pad (media->element, stream->srcpad);
1642 /* add stream now */
1643 g_array_append_val (media->streams, stream);
1645 setup_stream (stream, i, media);
1647 for (i = 0; i < 2; i++) {
1648 gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED);
1649 gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED);
1650 gst_element_set_state (stream->appqueue[i], GST_STATE_PAUSED);
1651 gst_element_set_state (stream->tee[i], GST_STATE_PAUSED);
1652 gst_element_set_state (stream->selector[i], GST_STATE_PAUSED);
1653 gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED);
1655 media->adding = FALSE;
1659 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1661 GST_INFO ("no more pads");
1662 if (media->fakesink) {
1663 gst_object_ref (media->fakesink);
1664 gst_bin_remove (GST_BIN (media->pipeline), media->fakesink);
1665 gst_element_set_state (media->fakesink, GST_STATE_NULL);
1666 gst_object_unref (media->fakesink);
1667 media->fakesink = NULL;
1668 GST_INFO ("removed fakesink");
1673 * gst_rtsp_media_prepare:
1674 * @media: a #GstRTSPMedia
1676 * Prepare @media for streaming. This function will create the pipeline and
1677 * other objects to manage the streaming.
1679 * It will preroll the pipeline and collect vital information about the streams
1680 * such as the duration.
1682 * Returns: %TRUE on success.
1685 gst_rtsp_media_prepare (GstRTSPMedia * media)
1687 GstStateChangeReturn ret;
1688 GstRTSPMediaStatus status;
1690 GstRTSPMediaClass *klass;
1694 if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1697 if (!media->reusable && media->reused)
1700 media->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1701 if (media->rtpbin == NULL)
1704 GST_INFO ("preparing media %p", media);
1706 /* reset some variables */
1707 media->is_live = FALSE;
1708 media->seekable = FALSE;
1709 media->buffering = FALSE;
1710 /* we're preparing now */
1711 media->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1713 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
1715 /* add the pipeline bus to our custom mainloop */
1716 media->source = gst_bus_create_watch (bus);
1717 gst_object_unref (bus);
1719 g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
1721 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1722 media->id = g_source_attach (media->source, klass->context);
1724 /* add stuff to the bin */
1725 gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
1727 /* link streams we already have, other streams might appear when we have
1728 * dynamic elements */
1729 n_streams = gst_rtsp_media_n_streams (media);
1730 for (i = 0; i < n_streams; i++) {
1731 GstRTSPMediaStream *stream;
1733 stream = gst_rtsp_media_get_stream (media, i);
1735 setup_stream (stream, i, media);
1738 for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
1739 GstElement *elem = walk->data;
1741 GST_INFO ("adding callbacks for dynamic element %p", elem);
1743 g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
1744 g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
1746 /* we add a fakesink here in order to make the state change async. We remove
1747 * the fakesink again in the no-more-pads callback. */
1748 media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1749 gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
1752 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1753 /* first go to PAUSED */
1754 ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1755 media->target_state = GST_STATE_PAUSED;
1758 case GST_STATE_CHANGE_SUCCESS:
1759 GST_INFO ("SUCCESS state change for media %p", media);
1760 media->seekable = TRUE;
1762 case GST_STATE_CHANGE_ASYNC:
1763 GST_INFO ("ASYNC state change for media %p", media);
1764 media->seekable = TRUE;
1766 case GST_STATE_CHANGE_NO_PREROLL:
1767 /* we need to go to PLAYING */
1768 GST_INFO ("NO_PREROLL state change: live media %p", media);
1769 /* FIXME we disable seeking for live streams for now. We should perform a
1770 * seeking query in preroll instead and do a seeking query. */
1771 media->seekable = FALSE;
1772 media->is_live = TRUE;
1773 ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1774 if (ret == GST_STATE_CHANGE_FAILURE)
1777 case GST_STATE_CHANGE_FAILURE:
1781 /* now wait for all pads to be prerolled */
1782 status = gst_rtsp_media_get_status (media);
1783 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1786 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1788 GST_INFO ("object %p is prerolled", media);
1800 GST_WARNING ("can not reuse media %p", media);
1805 GST_WARNING ("no rtpbin element");
1806 g_warning ("failed to create element 'rtpbin', check your installation");
1811 GST_WARNING ("failed to preroll pipeline");
1812 unlock_streams (media);
1813 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1814 gst_rtsp_media_unprepare (media);
1820 * gst_rtsp_media_unprepare:
1821 * @media: a #GstRTSPMedia
1823 * Unprepare @media. After this call, the media should be prepared again before
1824 * it can be used again. If the media is set to be non-reusable, a new instance
1827 * Returns: %TRUE on success.
1830 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1832 GstRTSPMediaClass *klass;
1835 if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1838 GST_INFO ("unprepare media %p", media);
1839 media->target_state = GST_STATE_NULL;
1841 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1842 if (klass->unprepare)
1843 success = klass->unprepare (media);
1847 media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1848 media->reused = TRUE;
1850 /* when the media is not reusable, this will effectively unref the media and
1852 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1858 default_unprepare (GstRTSPMedia * media)
1860 if (media->eos_shutdown) {
1861 GST_DEBUG ("sending EOS for shutdown");
1862 /* ref so that we don't disappear */
1863 g_object_ref (media);
1864 media->eos_pending = TRUE;
1865 gst_element_send_event (media->pipeline, gst_event_new_eos ());
1866 /* we need to go to playing again for the EOS to propagate, normally in this
1867 * state, nothing is receiving data from us anymore so this is ok. */
1868 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1870 GST_DEBUG ("shutting down");
1871 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1877 add_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
1878 gchar * dest, gint min, gint max)
1880 GST_INFO ("adding %s:%d-%d", dest, min, max);
1881 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1882 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1886 remove_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
1887 gchar * dest, gint min, gint max)
1889 GST_INFO ("removing %s:%d-%d", dest, min, max);
1890 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1891 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1895 set_multicast_ttl (GstRTSPMedia * media, GstRTSPMediaStream * stream, guint ttl)
1897 GST_INFO ("setting ttl-mc %d", ttl);
1898 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL);
1899 g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL);
1903 * gst_rtsp_media_set_state:
1904 * @media: a #GstRTSPMedia
1905 * @state: the target state of the media
1906 * @transports: a #GArray of #GstRTSPMediaTrans pointers
1908 * Set the state of @media to @state and for the transports in @transports.
1910 * Returns: %TRUE on success.
1913 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1914 GArray * transports)
1917 gboolean add, remove, do_state;
1920 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1921 g_return_val_if_fail (transports != NULL, FALSE);
1923 /* NULL and READY are the same */
1924 if (state == GST_STATE_READY)
1925 state = GST_STATE_NULL;
1927 add = remove = FALSE;
1929 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1933 case GST_STATE_NULL:
1934 /* unlock the streams so that they follow the state changes from now on */
1935 unlock_streams (media);
1937 case GST_STATE_PAUSED:
1938 /* we're going from PLAYING to PAUSED, READY or NULL, remove */
1939 if (media->target_state == GST_STATE_PLAYING)
1942 case GST_STATE_PLAYING:
1943 /* we're going to PLAYING, add */
1949 old_active = media->active;
1951 for (i = 0; i < transports->len; i++) {
1952 GstRTSPMediaTrans *tr;
1953 GstRTSPMediaStream *stream;
1954 GstRTSPTransport *trans;
1956 /* we need a non-NULL entry in the array */
1957 tr = g_array_index (transports, GstRTSPMediaTrans *, i);
1961 /* we need a transport */
1962 if (!(trans = tr->transport))
1965 /* get the stream and add the destinations */
1966 stream = gst_rtsp_media_get_stream (media, tr->idx);
1967 switch (trans->lower_transport) {
1968 case GST_RTSP_LOWER_TRANS_UDP:
1969 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1975 dest = trans->destination;
1976 if (trans->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1977 min = trans->port.min;
1978 max = trans->port.max;
1981 min = trans->client_port.min;
1982 max = trans->client_port.max;
1985 if (add && !tr->active) {
1986 add_udp_destination (media, stream, dest, min, max);
1988 set_multicast_ttl (media, stream, ttl);
1990 stream->transports = g_list_prepend (stream->transports, tr);
1993 } else if (remove && tr->active) {
1994 remove_udp_destination (media, stream, dest, min, max);
1995 stream->transports = g_list_remove (stream->transports, tr);
2001 case GST_RTSP_LOWER_TRANS_TCP:
2002 if (add && !tr->active) {
2003 GST_INFO ("adding TCP %s", trans->destination);
2004 stream->transports = g_list_prepend (stream->transports, tr);
2007 } else if (remove && tr->active) {
2008 GST_INFO ("removing TCP %s", trans->destination);
2009 stream->transports = g_list_remove (stream->transports, tr);
2015 GST_INFO ("Unknown transport %d", trans->lower_transport);
2020 /* we just added the first media, do the playing state change */
2021 if (old_active == 0 && add)
2023 /* if we have no more active media, do the downward state changes */
2024 else if (media->active == 0)
2029 GST_INFO ("state %d active %d media %p do_state %d", state, media->active,
2032 if (media->target_state != state) {
2034 if (state == GST_STATE_NULL) {
2035 gst_rtsp_media_unprepare (media);
2037 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
2039 media->target_state = state;
2040 gst_element_set_state (media->pipeline, state);
2043 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2047 /* remember where we are */
2048 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2049 old_active != media->active))
2050 collect_media_stats (media);
2056 * gst_rtsp_media_remove_elements:
2057 * @media: a #GstRTSPMedia
2059 * Remove all elements and the pipeline controlled by @media.
2062 gst_rtsp_media_remove_elements (GstRTSPMedia * media)
2066 unlock_streams (media);
2068 for (i = 0; i < media->streams->len; i++) {
2069 GstRTSPMediaStream *stream;
2071 GST_INFO ("Removing elements of stream %d from pipeline", i);
2073 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
2075 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
2077 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
2079 for (j = 0; j < 2; j++) {
2080 gst_element_set_state (stream->udpsrc[j], GST_STATE_NULL);
2081 gst_element_set_state (stream->udpsink[j], GST_STATE_NULL);
2082 gst_element_set_state (stream->appsrc[j], GST_STATE_NULL);
2083 gst_element_set_state (stream->appsink[j], GST_STATE_NULL);
2084 gst_element_set_state (stream->appqueue[j], GST_STATE_NULL);
2085 gst_element_set_state (stream->tee[j], GST_STATE_NULL);
2086 gst_element_set_state (stream->selector[j], GST_STATE_NULL);
2088 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsrc[j]);
2089 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsink[j]);
2090 gst_bin_remove (GST_BIN (media->pipeline), stream->appsrc[j]);
2091 gst_bin_remove (GST_BIN (media->pipeline), stream->appsink[j]);
2092 gst_bin_remove (GST_BIN (media->pipeline), stream->appqueue[j]);
2093 gst_bin_remove (GST_BIN (media->pipeline), stream->tee[j]);
2094 gst_bin_remove (GST_BIN (media->pipeline), stream->selector[j]);
2097 gst_caps_unref (stream->caps);
2098 stream->caps = NULL;
2099 gst_rtsp_media_stream_free (stream);
2101 g_array_remove_range (media->streams, 0, media->streams->len);
2103 gst_element_set_state (media->rtpbin, GST_STATE_NULL);
2104 gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
2106 gst_object_unref (media->pipeline);
2107 media->pipeline = NULL;
2111 default_handle_mtu (GstRTSPMedia * media, guint mtu)
2115 for (i = 0; i < media->streams->len; i++) {
2116 GstRTSPMediaStream *stream;
2118 GST_INFO ("Setting mtu %d for stream %d", mtu, i);
2120 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
2122 g_object_set (G_OBJECT (stream->payloader), "mtu", mtu, NULL);
2127 * gst_rtsp_media_handle_mtu:
2128 * @media: a #GstRTSPMedia
2131 * Set maximum size of one RTP packet on the payloaders.
2134 gst_rtsp_media_handle_mtu (GstRTSPMedia * media, guint mtu)
2136 GstRTSPMediaClass *klass;
2138 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2140 if (klass->handle_mtu)
2141 klass->handle_mtu (media, mtu);