2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: The media pipeline
24 * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
25 * #GstRTSPSessionMedia
27 * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
28 * streaming to the clients. The actual data transfer is done by the
29 * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
31 * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
32 * client does a DESCRIBE or SETUP of a resource.
34 * A media is created with gst_rtsp_media_new() that takes the element that will
35 * provide the streaming elements. For each of the streams, a new #GstRTSPStream
36 * object needs to be made with the gst_rtsp_media_create_stream() which takes
37 * the payloader element and the source pad that produces the RTP stream.
39 * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
40 * prepare method will add rtpbin and sinks and sources to send and receive RTP
41 * and RTCP packets from the clients. Each stream srcpad is connected to an
42 * input into the internal rtpbin.
44 * It is also possible to dynamically create #GstRTSPStream objects during the
45 * prepare phase. With gst_rtsp_media_get_status() you can check the status of
48 * After the media is prepared, it is ready for streaming. It will usually be
49 * managed in a session with gst_rtsp_session_manage_media(). See
50 * #GstRTSPSession and #GstRTSPSessionMedia.
52 * The state of the media can be controlled with gst_rtsp_media_set_state ().
53 * Seeking can be done with gst_rtsp_media_seek().
55 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
56 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
59 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
60 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
61 * can be prepared again after an unprepare.
63 * Last reviewed on 2013-07-11 (1.0.0)
70 #include <gst/app/gstappsrc.h>
71 #include <gst/app/gstappsink.h>
73 #include <gst/sdp/gstmikey.h>
74 #include <gst/rtp/gstrtppayloads.h>
76 #define AES_128_KEY_LEN 16
77 #define AES_256_KEY_LEN 32
79 #define HMAC_32_KEY_LEN 4
80 #define HMAC_80_KEY_LEN 10
82 #include "rtsp-media.h"
84 #define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
85 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
87 struct _GstRTSPMediaPrivate
92 /* protected by lock */
93 GstRTSPPermissions *permissions;
95 gboolean suspend_mode;
97 GstRTSPProfile profiles;
98 GstRTSPLowerTrans protocols;
100 gboolean eos_shutdown;
102 GstRTSPAddressPool *pool;
104 GstRTSPTransportMode transport_mode;
107 GRecMutex state_lock; /* locking order: state lock, lock */
108 GPtrArray *streams; /* protected by lock */
109 GList *dynamic; /* protected by lock */
110 GstRTSPMediaStatus status; /* protected by lock */
115 /* the pipeline for the media */
116 GstElement *pipeline;
117 GstElement *fakesink; /* protected by lock */
120 GstRTSPThread *thread;
122 gboolean time_provider;
123 GstNetTimeProvider *nettime;
128 GstState target_state;
130 /* RTP session manager */
133 /* the range of media */
134 GstRTSPTimeRange range; /* protected by lock */
135 GstClockTime range_start;
136 GstClockTime range_stop;
138 GList *payloads; /* protected by lock */
139 GstClockTime rtx_time; /* protected by lock */
140 guint latency; /* protected by lock */
143 #define DEFAULT_SHARED FALSE
144 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
145 #define DEFAULT_REUSABLE FALSE
146 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
147 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
148 GST_RTSP_LOWER_TRANS_TCP
149 #define DEFAULT_EOS_SHUTDOWN FALSE
150 #define DEFAULT_BUFFER_SIZE 0x80000
151 #define DEFAULT_TIME_PROVIDER FALSE
152 #define DEFAULT_LATENCY 200
153 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
155 /* define to dump received RTCP packets */
178 SIGNAL_REMOVED_STREAM,
186 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
187 #define GST_CAT_DEFAULT rtsp_media_debug
189 static void gst_rtsp_media_get_property (GObject * object, guint propid,
190 GValue * value, GParamSpec * pspec);
191 static void gst_rtsp_media_set_property (GObject * object, guint propid,
192 const GValue * value, GParamSpec * pspec);
193 static void gst_rtsp_media_finalize (GObject * obj);
195 static gboolean default_handle_message (GstRTSPMedia * media,
196 GstMessage * message);
197 static void finish_unprepare (GstRTSPMedia * media);
198 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
199 static gboolean default_unprepare (GstRTSPMedia * media);
200 static gboolean default_suspend (GstRTSPMedia * media);
201 static gboolean default_unsuspend (GstRTSPMedia * media);
202 static gboolean default_convert_range (GstRTSPMedia * media,
203 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
204 static gboolean default_query_position (GstRTSPMedia * media,
206 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
207 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
208 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
210 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
212 static gboolean wait_preroll (GstRTSPMedia * media);
214 static GstElement * find_payload_element (GstElement * payloader);
216 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
218 #define C_ENUM(v) ((gint) v)
221 gst_rtsp_suspend_mode_get_type (void)
224 static const GEnumValue values[] = {
225 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
226 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
228 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
233 if (g_once_init_enter (&id)) {
234 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
235 g_once_init_leave (&id, tmp);
240 #define C_FLAGS(v) ((guint) v)
243 gst_rtsp_transport_mode_get_type (void)
246 static const GFlagsValue values[] = {
247 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
249 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
254 if (g_once_init_enter (&id)) {
255 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
256 g_once_init_leave (&id, tmp);
261 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
264 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
266 GObjectClass *gobject_class;
268 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
270 gobject_class = G_OBJECT_CLASS (klass);
272 gobject_class->get_property = gst_rtsp_media_get_property;
273 gobject_class->set_property = gst_rtsp_media_set_property;
274 gobject_class->finalize = gst_rtsp_media_finalize;
276 g_object_class_install_property (gobject_class, PROP_SHARED,
277 g_param_spec_boolean ("shared", "Shared",
278 "If this media pipeline can be shared", DEFAULT_SHARED,
279 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
281 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
282 g_param_spec_enum ("suspend-mode", "Suspend Mode",
283 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
284 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
286 g_object_class_install_property (gobject_class, PROP_REUSABLE,
287 g_param_spec_boolean ("reusable", "Reusable",
288 "If this media pipeline can be reused after an unprepare",
289 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
291 g_object_class_install_property (gobject_class, PROP_PROFILES,
292 g_param_spec_flags ("profiles", "Profiles",
293 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
294 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
296 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
297 g_param_spec_flags ("protocols", "Protocols",
298 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
299 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
301 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
302 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
303 "Send an EOS event to the pipeline before unpreparing",
304 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
306 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
307 g_param_spec_uint ("buffer-size", "Buffer Size",
308 "The kernel UDP buffer size to use", 0, G_MAXUINT,
309 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
311 g_object_class_install_property (gobject_class, PROP_ELEMENT,
312 g_param_spec_object ("element", "The Element",
313 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
314 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
316 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
317 g_param_spec_boolean ("time-provider", "Time Provider",
318 "Use a NetTimeProvider for clients",
319 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
321 g_object_class_install_property (gobject_class, PROP_LATENCY,
322 g_param_spec_uint ("latency", "Latency",
323 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
324 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
326 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
327 g_param_spec_flags ("transport-mode", "Transport Mode",
328 "If this media pipeline can be used for PLAY or RECORD",
329 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
330 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
332 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
333 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
334 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
335 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
337 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
338 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
339 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
340 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
341 GST_TYPE_RTSP_STREAM);
343 gst_rtsp_media_signals[SIGNAL_PREPARED] =
344 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
345 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
346 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
348 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
349 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
350 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
351 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
353 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
354 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
355 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
356 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
358 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
359 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
360 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
361 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
363 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
365 klass->handle_message = default_handle_message;
366 klass->prepare = default_prepare;
367 klass->unprepare = default_unprepare;
368 klass->suspend = default_suspend;
369 klass->unsuspend = default_unsuspend;
370 klass->convert_range = default_convert_range;
371 klass->query_position = default_query_position;
372 klass->query_stop = default_query_stop;
373 klass->create_rtpbin = default_create_rtpbin;
374 klass->setup_sdp = default_setup_sdp;
375 klass->handle_sdp = default_handle_sdp;
379 gst_rtsp_media_init (GstRTSPMedia * media)
381 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
385 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
386 g_mutex_init (&priv->lock);
387 g_cond_init (&priv->cond);
388 g_rec_mutex_init (&priv->state_lock);
390 priv->shared = DEFAULT_SHARED;
391 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
392 priv->reusable = DEFAULT_REUSABLE;
393 priv->profiles = DEFAULT_PROFILES;
394 priv->protocols = DEFAULT_PROTOCOLS;
395 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
396 priv->buffer_size = DEFAULT_BUFFER_SIZE;
397 priv->time_provider = DEFAULT_TIME_PROVIDER;
398 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
402 gst_rtsp_media_finalize (GObject * obj)
404 GstRTSPMediaPrivate *priv;
407 media = GST_RTSP_MEDIA (obj);
410 GST_INFO ("finalize media %p", media);
412 if (priv->permissions)
413 gst_rtsp_permissions_unref (priv->permissions);
415 g_ptr_array_unref (priv->streams);
417 g_list_free_full (priv->dynamic, gst_object_unref);
420 gst_object_unref (priv->pipeline);
422 gst_object_unref (priv->nettime);
423 gst_object_unref (priv->element);
425 g_object_unref (priv->pool);
427 g_list_free (priv->payloads);
428 g_mutex_clear (&priv->lock);
429 g_cond_clear (&priv->cond);
430 g_rec_mutex_clear (&priv->state_lock);
432 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
436 gst_rtsp_media_get_property (GObject * object, guint propid,
437 GValue * value, GParamSpec * pspec)
439 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
443 g_value_set_object (value, media->priv->element);
446 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
448 case PROP_SUSPEND_MODE:
449 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
452 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
455 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
458 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
460 case PROP_EOS_SHUTDOWN:
461 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
463 case PROP_BUFFER_SIZE:
464 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
466 case PROP_TIME_PROVIDER:
467 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
470 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
472 case PROP_TRANSPORT_MODE:
473 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
476 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
481 gst_rtsp_media_set_property (GObject * object, guint propid,
482 const GValue * value, GParamSpec * pspec)
484 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
488 media->priv->element = g_value_get_object (value);
489 gst_object_ref_sink (media->priv->element);
492 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
494 case PROP_SUSPEND_MODE:
495 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
498 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
501 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
504 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
506 case PROP_EOS_SHUTDOWN:
507 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
509 case PROP_BUFFER_SIZE:
510 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
512 case PROP_TIME_PROVIDER:
513 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
516 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
518 case PROP_TRANSPORT_MODE:
519 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
522 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
530 } DoQueryPositionData;
533 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
537 if (gst_rtsp_stream_query_position (stream, &tmp)) {
538 data->position = MAX (data->position, tmp);
544 default_query_position (GstRTSPMedia * media, gint64 * position)
546 GstRTSPMediaPrivate *priv;
547 DoQueryPositionData data;
554 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
556 *position = data.position;
568 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
572 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
573 data->stop = MAX (data->stop, tmp);
579 default_query_stop (GstRTSPMedia * media, gint64 * stop)
581 GstRTSPMediaPrivate *priv;
582 DoQueryStopData data;
589 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
597 default_create_rtpbin (GstRTSPMedia * media)
601 rtpbin = gst_element_factory_make ("rtpbin", NULL);
606 /* must be called with state lock */
608 collect_media_stats (GstRTSPMedia * media)
610 GstRTSPMediaPrivate *priv = media->priv;
611 gint64 position = 0, stop = -1;
613 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
614 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
617 priv->range.unit = GST_RTSP_RANGE_NPT;
619 GST_INFO ("collect media stats");
622 priv->range.min.type = GST_RTSP_TIME_NOW;
623 priv->range.min.seconds = -1;
624 priv->range_start = -1;
625 priv->range.max.type = GST_RTSP_TIME_END;
626 priv->range.max.seconds = -1;
627 priv->range_stop = -1;
629 GstRTSPMediaClass *klass;
632 klass = GST_RTSP_MEDIA_GET_CLASS (media);
634 /* get the position */
636 if (klass->query_position)
637 ret = klass->query_position (media, &position);
640 GST_INFO ("position query failed");
644 /* get the current segment stop */
646 if (klass->query_stop)
647 ret = klass->query_stop (media, &stop);
650 GST_INFO ("stop query failed");
654 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
655 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
657 if (position == -1) {
658 priv->range.min.type = GST_RTSP_TIME_NOW;
659 priv->range.min.seconds = -1;
660 priv->range_start = -1;
662 priv->range.min.type = GST_RTSP_TIME_SECONDS;
663 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
664 priv->range_start = position;
667 priv->range.max.type = GST_RTSP_TIME_END;
668 priv->range.max.seconds = -1;
669 priv->range_stop = -1;
671 priv->range.max.type = GST_RTSP_TIME_SECONDS;
672 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
673 priv->range_stop = stop;
679 * gst_rtsp_media_new:
680 * @element: (transfer full): a #GstElement
682 * Create a new #GstRTSPMedia instance. @element is the bin element that
683 * provides the different streams. The #GstRTSPMedia object contains the
684 * element to produce RTP data for one or more related (audio/video/..)
687 * Ownership is taken of @element.
689 * Returns: (transfer full): a new #GstRTSPMedia object.
692 gst_rtsp_media_new (GstElement * element)
694 GstRTSPMedia *result;
696 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
698 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
704 * gst_rtsp_media_get_element:
705 * @media: a #GstRTSPMedia
707 * Get the element that was used when constructing @media.
709 * Returns: (transfer full): a #GstElement. Unref after usage.
712 gst_rtsp_media_get_element (GstRTSPMedia * media)
714 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
716 return gst_object_ref (media->priv->element);
720 * gst_rtsp_media_take_pipeline:
721 * @media: a #GstRTSPMedia
722 * @pipeline: (transfer full): a #GstPipeline
724 * Set @pipeline as the #GstPipeline for @media. Ownership is
725 * taken of @pipeline.
728 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
730 GstRTSPMediaPrivate *priv;
732 GstNetTimeProvider *nettime;
734 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
735 g_return_if_fail (GST_IS_PIPELINE (pipeline));
739 g_mutex_lock (&priv->lock);
740 old = priv->pipeline;
741 priv->pipeline = GST_ELEMENT_CAST (pipeline);
742 nettime = priv->nettime;
743 priv->nettime = NULL;
744 g_mutex_unlock (&priv->lock);
747 gst_object_unref (old);
750 gst_object_unref (nettime);
752 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
756 * gst_rtsp_media_set_permissions:
757 * @media: a #GstRTSPMedia
758 * @permissions: (transfer none): a #GstRTSPPermissions
760 * Set @permissions on @media.
763 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
764 GstRTSPPermissions * permissions)
766 GstRTSPMediaPrivate *priv;
768 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
772 g_mutex_lock (&priv->lock);
773 if (priv->permissions)
774 gst_rtsp_permissions_unref (priv->permissions);
775 if ((priv->permissions = permissions))
776 gst_rtsp_permissions_ref (permissions);
777 g_mutex_unlock (&priv->lock);
781 * gst_rtsp_media_get_permissions:
782 * @media: a #GstRTSPMedia
784 * Get the permissions object from @media.
786 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
789 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
791 GstRTSPMediaPrivate *priv;
792 GstRTSPPermissions *result;
794 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
798 g_mutex_lock (&priv->lock);
799 if ((result = priv->permissions))
800 gst_rtsp_permissions_ref (result);
801 g_mutex_unlock (&priv->lock);
807 * gst_rtsp_media_set_suspend_mode:
808 * @media: a #GstRTSPMedia
809 * @mode: the new #GstRTSPSuspendMode
811 * Control how @ media will be suspended after the SDP has been generated and
812 * after a PAUSE request has been performed.
814 * Media must be unprepared when setting the suspend mode.
817 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
819 GstRTSPMediaPrivate *priv;
821 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
825 g_rec_mutex_lock (&priv->state_lock);
826 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
828 priv->suspend_mode = mode;
829 g_rec_mutex_unlock (&priv->state_lock);
836 GST_WARNING ("media %p was prepared", media);
837 g_rec_mutex_unlock (&priv->state_lock);
842 * gst_rtsp_media_get_suspend_mode:
843 * @media: a #GstRTSPMedia
845 * Get how @media will be suspended.
847 * Returns: #GstRTSPSuspendMode.
850 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
852 GstRTSPMediaPrivate *priv;
853 GstRTSPSuspendMode res;
855 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
859 g_rec_mutex_lock (&priv->state_lock);
860 res = priv->suspend_mode;
861 g_rec_mutex_unlock (&priv->state_lock);
867 * gst_rtsp_media_set_shared:
868 * @media: a #GstRTSPMedia
869 * @shared: the new value
871 * Set or unset if the pipeline for @media can be shared will multiple clients.
872 * When @shared is %TRUE, client requests for this media will share the media
876 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
878 GstRTSPMediaPrivate *priv;
880 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
884 g_mutex_lock (&priv->lock);
885 priv->shared = shared;
886 g_mutex_unlock (&priv->lock);
890 * gst_rtsp_media_is_shared:
891 * @media: a #GstRTSPMedia
893 * Check if the pipeline for @media can be shared between multiple clients.
895 * Returns: %TRUE if the media can be shared between clients.
898 gst_rtsp_media_is_shared (GstRTSPMedia * media)
900 GstRTSPMediaPrivate *priv;
903 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
907 g_mutex_lock (&priv->lock);
909 g_mutex_unlock (&priv->lock);
915 * gst_rtsp_media_set_reusable:
916 * @media: a #GstRTSPMedia
917 * @reusable: the new value
919 * Set or unset if the pipeline for @media can be reused after the pipeline has
923 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
925 GstRTSPMediaPrivate *priv;
927 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
931 g_mutex_lock (&priv->lock);
932 priv->reusable = reusable;
933 g_mutex_unlock (&priv->lock);
937 * gst_rtsp_media_is_reusable:
938 * @media: a #GstRTSPMedia
940 * Check if the pipeline for @media can be reused after an unprepare.
942 * Returns: %TRUE if the media can be reused
945 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
947 GstRTSPMediaPrivate *priv;
950 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
954 g_mutex_lock (&priv->lock);
955 res = priv->reusable;
956 g_mutex_unlock (&priv->lock);
962 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
964 gst_rtsp_stream_set_profiles (stream, *profiles);
968 * gst_rtsp_media_set_profiles:
969 * @media: a #GstRTSPMedia
970 * @profiles: the new flags
972 * Configure the allowed lower transport for @media.
975 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
977 GstRTSPMediaPrivate *priv;
979 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
983 g_mutex_lock (&priv->lock);
984 priv->profiles = profiles;
985 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
986 g_mutex_unlock (&priv->lock);
990 * gst_rtsp_media_get_profiles:
991 * @media: a #GstRTSPMedia
993 * Get the allowed profiles of @media.
995 * Returns: a #GstRTSPProfile
998 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1000 GstRTSPMediaPrivate *priv;
1003 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1007 g_mutex_lock (&priv->lock);
1008 res = priv->profiles;
1009 g_mutex_unlock (&priv->lock);
1015 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1017 gst_rtsp_stream_set_protocols (stream, *protocols);
1021 * gst_rtsp_media_set_protocols:
1022 * @media: a #GstRTSPMedia
1023 * @protocols: the new flags
1025 * Configure the allowed lower transport for @media.
1028 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1030 GstRTSPMediaPrivate *priv;
1032 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1036 g_mutex_lock (&priv->lock);
1037 priv->protocols = protocols;
1038 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1039 g_mutex_unlock (&priv->lock);
1043 * gst_rtsp_media_get_protocols:
1044 * @media: a #GstRTSPMedia
1046 * Get the allowed protocols of @media.
1048 * Returns: a #GstRTSPLowerTrans
1051 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1053 GstRTSPMediaPrivate *priv;
1054 GstRTSPLowerTrans res;
1056 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1057 GST_RTSP_LOWER_TRANS_UNKNOWN);
1061 g_mutex_lock (&priv->lock);
1062 res = priv->protocols;
1063 g_mutex_unlock (&priv->lock);
1069 * gst_rtsp_media_set_eos_shutdown:
1070 * @media: a #GstRTSPMedia
1071 * @eos_shutdown: the new value
1073 * Set or unset if an EOS event will be sent to the pipeline for @media before
1077 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1079 GstRTSPMediaPrivate *priv;
1081 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1085 g_mutex_lock (&priv->lock);
1086 priv->eos_shutdown = eos_shutdown;
1087 g_mutex_unlock (&priv->lock);
1091 * gst_rtsp_media_is_eos_shutdown:
1092 * @media: a #GstRTSPMedia
1094 * Check if the pipeline for @media will send an EOS down the pipeline before
1097 * Returns: %TRUE if the media will send EOS before unpreparing.
1100 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1102 GstRTSPMediaPrivate *priv;
1105 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1109 g_mutex_lock (&priv->lock);
1110 res = priv->eos_shutdown;
1111 g_mutex_unlock (&priv->lock);
1117 * gst_rtsp_media_set_buffer_size:
1118 * @media: a #GstRTSPMedia
1119 * @size: the new value
1121 * Set the kernel UDP buffer size.
1124 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1126 GstRTSPMediaPrivate *priv;
1128 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1130 GST_LOG_OBJECT (media, "set buffer size %u", size);
1134 g_mutex_lock (&priv->lock);
1135 priv->buffer_size = size;
1136 g_mutex_unlock (&priv->lock);
1140 * gst_rtsp_media_get_buffer_size:
1141 * @media: a #GstRTSPMedia
1143 * Get the kernel UDP buffer size.
1145 * Returns: the kernel UDP buffer size.
1148 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1150 GstRTSPMediaPrivate *priv;
1153 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1157 g_mutex_lock (&priv->lock);
1158 res = priv->buffer_size;
1159 g_mutex_unlock (&priv->lock);
1165 * gst_rtsp_media_set_retransmission_time:
1166 * @media: a #GstRTSPMedia
1167 * @time: the new value
1169 * Set the amount of time to store retransmission packets.
1172 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1174 GstRTSPMediaPrivate *priv;
1177 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1179 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1183 g_mutex_lock (&priv->lock);
1184 priv->rtx_time = time;
1185 for (i = 0; i < priv->streams->len; i++) {
1186 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1188 gst_rtsp_stream_set_retransmission_time (stream, time);
1192 g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
1193 g_mutex_unlock (&priv->lock);
1197 * gst_rtsp_media_get_retransmission_time:
1198 * @media: a #GstRTSPMedia
1200 * Get the amount of time to store retransmission data.
1202 * Returns: the amount of time to store retransmission data.
1205 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1207 GstRTSPMediaPrivate *priv;
1210 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1214 g_mutex_unlock (&priv->lock);
1215 res = priv->rtx_time;
1216 g_mutex_unlock (&priv->lock);
1222 * gst_rtsp_media_set_latncy:
1223 * @media: a #GstRTSPMedia
1224 * @latency: latency in milliseconds
1226 * Configure the latency used for receiving media.
1229 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1231 GstRTSPMediaPrivate *priv;
1233 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1235 GST_LOG_OBJECT (media, "set latency %ums", latency);
1239 g_mutex_lock (&priv->lock);
1240 priv->latency = latency;
1242 g_object_set (priv->rtpbin, "latency", latency, NULL);
1243 g_mutex_unlock (&priv->lock);
1247 * gst_rtsp_media_get_latency:
1248 * @media: a #GstRTSPMedia
1250 * Get the latency that is used for receiving media.
1252 * Returns: latency in milliseconds
1255 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1257 GstRTSPMediaPrivate *priv;
1260 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1264 g_mutex_unlock (&priv->lock);
1265 res = priv->latency;
1266 g_mutex_unlock (&priv->lock);
1272 * gst_rtsp_media_use_time_provider:
1273 * @media: a #GstRTSPMedia
1274 * @time_provider: if a #GstNetTimeProvider should be used
1276 * Set @media to provide a #GstNetTimeProvider.
1279 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1281 GstRTSPMediaPrivate *priv;
1283 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1287 g_mutex_lock (&priv->lock);
1288 priv->time_provider = time_provider;
1289 g_mutex_unlock (&priv->lock);
1293 * gst_rtsp_media_is_time_provider:
1294 * @media: a #GstRTSPMedia
1296 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1298 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1300 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1303 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1305 GstRTSPMediaPrivate *priv;
1308 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1312 g_mutex_unlock (&priv->lock);
1313 res = priv->time_provider;
1314 g_mutex_unlock (&priv->lock);
1320 * gst_rtsp_media_set_address_pool:
1321 * @media: a #GstRTSPMedia
1322 * @pool: (transfer none): a #GstRTSPAddressPool
1324 * configure @pool to be used as the address pool of @media.
1327 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1328 GstRTSPAddressPool * pool)
1330 GstRTSPMediaPrivate *priv;
1331 GstRTSPAddressPool *old;
1333 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1337 GST_LOG_OBJECT (media, "set address pool %p", pool);
1339 g_mutex_lock (&priv->lock);
1340 if ((old = priv->pool) != pool)
1341 priv->pool = pool ? g_object_ref (pool) : NULL;
1344 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1346 g_mutex_unlock (&priv->lock);
1349 g_object_unref (old);
1353 * gst_rtsp_media_get_address_pool:
1354 * @media: a #GstRTSPMedia
1356 * Get the #GstRTSPAddressPool used as the address pool of @media.
1358 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1361 GstRTSPAddressPool *
1362 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1364 GstRTSPMediaPrivate *priv;
1365 GstRTSPAddressPool *result;
1367 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1371 g_mutex_lock (&priv->lock);
1372 if ((result = priv->pool))
1373 g_object_ref (result);
1374 g_mutex_unlock (&priv->lock);
1380 _find_payload_types (GstRTSPMedia * media)
1383 GQueue queue = G_QUEUE_INIT;
1385 n = media->priv->streams->len;
1386 for (i = 0; i < n; i++) {
1387 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1388 guint pt = gst_rtsp_stream_get_pt (stream);
1390 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1397 _next_available_pt (GList * payloads)
1401 for (i = 96; i <= 127; i++) {
1402 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1404 return GPOINTER_TO_UINT (i);
1411 * gst_rtsp_media_collect_streams:
1412 * @media: a #GstRTSPMedia
1414 * Find all payloader elements, they should be named pay\%d in the
1415 * element of @media, and create #GstRTSPStreams for them.
1417 * Collect all dynamic elements, named dynpay\%d, and add them to
1418 * the list of dynamic elements.
1420 * Find all depayloader elements, they should be named depay\%d in the
1421 * element of @media, and create #GstRTSPStreams for them.
1424 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1426 GstRTSPMediaPrivate *priv;
1427 GstElement *element, *elem;
1431 gboolean more_elem_remaining = TRUE;
1432 GstRTSPTransportMode mode = 0;
1434 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1437 element = priv->element;
1440 for (i = 0; more_elem_remaining; i++) {
1443 more_elem_remaining = FALSE;
1445 name = g_strdup_printf ("pay%d", i);
1446 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1448 GST_INFO ("found stream %d with payloader %p", i, elem);
1450 /* take the pad of the payloader */
1451 pad = gst_element_get_static_pad (elem, "src");
1453 /* find the real payload element in case elem is a GstBin */
1454 pay = find_payload_element (elem);
1456 /* create the stream */
1458 GST_WARNING ("could not find real payloader, using bin");
1459 gst_rtsp_media_create_stream (media, elem, pad);
1461 gst_rtsp_media_create_stream (media, pay, pad);
1462 gst_object_unref (pay);
1465 gst_object_unref (pad);
1466 gst_object_unref (elem);
1469 more_elem_remaining = TRUE;
1470 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1474 name = g_strdup_printf ("dynpay%d", i);
1475 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1476 /* a stream that will dynamically create pads to provide RTP packets */
1477 GST_INFO ("found dynamic element %d, %p", i, elem);
1479 g_mutex_lock (&priv->lock);
1480 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1481 g_mutex_unlock (&priv->lock);
1484 more_elem_remaining = TRUE;
1485 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1489 name = g_strdup_printf ("depay%d", i);
1490 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1491 GST_INFO ("found stream %d with depayloader %p", i, elem);
1493 /* take the pad of the payloader */
1494 pad = gst_element_get_static_pad (elem, "sink");
1495 /* create the stream */
1496 gst_rtsp_media_create_stream (media, elem, pad);
1497 gst_object_unref (pad);
1498 gst_object_unref (elem);
1501 more_elem_remaining = TRUE;
1502 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1508 if (priv->transport_mode != mode)
1509 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1510 priv->transport_mode, mode);
1515 * gst_rtsp_media_create_stream:
1516 * @media: a #GstRTSPMedia
1517 * @payloader: a #GstElement
1520 * Create a new stream in @media that provides RTP data on @pad.
1521 * @pad should be a pad of an element inside @media->element.
1523 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1527 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1530 GstRTSPMediaPrivate *priv;
1531 GstRTSPStream *stream;
1536 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1537 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1538 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1542 g_mutex_lock (&priv->lock);
1543 idx = priv->streams->len;
1545 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1547 if (GST_PAD_IS_SRC (pad))
1548 name = g_strdup_printf ("src_%u", idx);
1550 name = g_strdup_printf ("sink_%u", idx);
1552 ghostpad = gst_ghost_pad_new (name, pad);
1553 gst_pad_set_active (ghostpad, TRUE);
1554 gst_element_add_pad (priv->element, ghostpad);
1557 stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
1559 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1560 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1561 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1562 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
1564 g_ptr_array_add (priv->streams, stream);
1566 if (GST_PAD_IS_SRC (pad)) {
1570 g_list_free (priv->payloads);
1571 priv->payloads = _find_payload_types (media);
1573 n = priv->streams->len;
1574 for (i = 0; i < n; i++) {
1575 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1576 guint rtx_pt = _next_available_pt (priv->payloads);
1579 GST_WARNING ("Ran out of space of dynamic payload types");
1583 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
1586 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
1589 g_mutex_unlock (&priv->lock);
1591 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1598 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1600 GstRTSPMediaPrivate *priv;
1605 g_mutex_lock (&priv->lock);
1606 /* remove the ghostpad */
1607 srcpad = gst_rtsp_stream_get_srcpad (stream);
1608 gst_element_remove_pad (priv->element, srcpad);
1609 gst_object_unref (srcpad);
1610 /* now remove the stream */
1611 g_object_ref (stream);
1612 g_ptr_array_remove (priv->streams, stream);
1613 g_mutex_unlock (&priv->lock);
1615 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1618 g_object_unref (stream);
1622 * gst_rtsp_media_n_streams:
1623 * @media: a #GstRTSPMedia
1625 * Get the number of streams in this media.
1627 * Returns: The number of streams.
1630 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1632 GstRTSPMediaPrivate *priv;
1635 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1639 g_mutex_lock (&priv->lock);
1640 res = priv->streams->len;
1641 g_mutex_unlock (&priv->lock);
1647 * gst_rtsp_media_get_stream:
1648 * @media: a #GstRTSPMedia
1649 * @idx: the stream index
1651 * Retrieve the stream with index @idx from @media.
1653 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
1654 * @idx or %NULL when a stream with that index did not exist.
1657 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1659 GstRTSPMediaPrivate *priv;
1662 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1666 g_mutex_lock (&priv->lock);
1667 if (idx < priv->streams->len)
1668 res = g_ptr_array_index (priv->streams, idx);
1671 g_mutex_unlock (&priv->lock);
1677 * gst_rtsp_media_find_stream:
1678 * @media: a #GstRTSPMedia
1679 * @control: the control of the stream
1681 * Find a stream in @media with @control as the control uri.
1683 * Returns: (nullable) (transfer none): the #GstRTSPStream with
1684 * control uri @control or %NULL when a stream with that control did
1688 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1690 GstRTSPMediaPrivate *priv;
1694 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1695 g_return_val_if_fail (control != NULL, NULL);
1701 g_mutex_lock (&priv->lock);
1702 for (i = 0; i < priv->streams->len; i++) {
1703 GstRTSPStream *test;
1705 test = g_ptr_array_index (priv->streams, i);
1706 if (gst_rtsp_stream_has_control (test, control)) {
1711 g_mutex_unlock (&priv->lock);
1716 /* called with state-lock */
1718 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1719 GstRTSPRangeUnit unit)
1721 return gst_rtsp_range_convert_units (range, unit);
1725 * gst_rtsp_media_get_range_string:
1726 * @media: a #GstRTSPMedia
1727 * @play: for the PLAY request
1728 * @unit: the unit to use for the string
1730 * Get the current range as a string. @media must be prepared with
1731 * gst_rtsp_media_prepare ().
1733 * Returns: (transfer full): The range as a string, g_free() after usage.
1736 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1737 GstRTSPRangeUnit unit)
1739 GstRTSPMediaClass *klass;
1740 GstRTSPMediaPrivate *priv;
1742 GstRTSPTimeRange range;
1744 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1745 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1746 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1750 g_rec_mutex_lock (&priv->state_lock);
1751 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1752 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1755 g_mutex_lock (&priv->lock);
1757 /* Update the range value with current position/duration */
1758 collect_media_stats (media);
1761 range = priv->range;
1763 if (!play && priv->n_active > 0) {
1764 range.min.type = GST_RTSP_TIME_NOW;
1765 range.min.seconds = -1;
1767 g_mutex_unlock (&priv->lock);
1768 g_rec_mutex_unlock (&priv->state_lock);
1770 if (!klass->convert_range (media, &range, unit))
1771 goto conversion_failed;
1773 result = gst_rtsp_range_to_string (&range);
1780 GST_WARNING ("media %p was not prepared", media);
1781 g_rec_mutex_unlock (&priv->state_lock);
1786 GST_WARNING ("range conversion to unit %d failed", unit);
1792 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1794 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1798 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1800 GstRTSPMediaPrivate *priv = media->priv;
1802 GST_DEBUG ("media %p set blocked %d", media, blocked);
1803 priv->blocked = blocked;
1804 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1808 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1810 GstRTSPMediaPrivate *priv = media->priv;
1812 g_mutex_lock (&priv->lock);
1813 priv->status = status;
1814 GST_DEBUG ("setting new status to %d", status);
1815 g_cond_broadcast (&priv->cond);
1816 g_mutex_unlock (&priv->lock);
1820 * gst_rtsp_media_get_status:
1821 * @media: a #GstRTSPMedia
1823 * Get the status of @media. When @media is busy preparing, this function waits
1824 * until @media is prepared or in error.
1826 * Returns: the status of @media.
1829 gst_rtsp_media_get_status (GstRTSPMedia * media)
1831 GstRTSPMediaPrivate *priv = media->priv;
1832 GstRTSPMediaStatus result;
1835 g_mutex_lock (&priv->lock);
1836 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1837 /* while we are preparing, wait */
1838 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1839 GST_DEBUG ("waiting for status change");
1840 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1841 GST_DEBUG ("timeout, assuming error status");
1842 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1845 /* could be success or error */
1846 result = priv->status;
1847 GST_DEBUG ("got status %d", result);
1848 g_mutex_unlock (&priv->lock);
1854 * gst_rtsp_media_seek:
1855 * @media: a #GstRTSPMedia
1856 * @range: (transfer none): a #GstRTSPTimeRange
1858 * Seek the pipeline of @media to @range. @media must be prepared with
1859 * gst_rtsp_media_prepare().
1861 * Returns: %TRUE on success.
1864 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1866 GstRTSPMediaClass *klass;
1867 GstRTSPMediaPrivate *priv;
1869 GstClockTime start, stop;
1870 GstSeekType start_type, stop_type;
1872 gint64 current_position;
1874 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1876 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1877 g_return_val_if_fail (range != NULL, FALSE);
1878 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1882 g_rec_mutex_lock (&priv->state_lock);
1883 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1886 /* Update the seekable state of the pipeline in case it changed */
1887 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
1888 /* TODO: Seeking for RECORD? */
1889 priv->seekable = FALSE;
1891 query = gst_query_new_seeking (GST_FORMAT_TIME);
1892 if (gst_element_query (priv->pipeline, query)) {
1897 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1898 priv->seekable = seekable;
1900 gst_query_unref (query);
1903 if (!priv->seekable)
1906 start_type = stop_type = GST_SEEK_TYPE_NONE;
1908 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1910 gst_rtsp_range_get_times (range, &start, &stop);
1912 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1913 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1914 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1915 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1917 current_position = -1;
1918 if (klass->query_position)
1919 klass->query_position (media, ¤t_position);
1920 GST_INFO ("current media position %" GST_TIME_FORMAT,
1921 GST_TIME_ARGS (current_position));
1923 if (start != GST_CLOCK_TIME_NONE)
1924 start_type = GST_SEEK_TYPE_SET;
1926 if (priv->range_stop == stop)
1927 stop = GST_CLOCK_TIME_NONE;
1928 else if (stop != GST_CLOCK_TIME_NONE)
1929 stop_type = GST_SEEK_TYPE_SET;
1931 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1934 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1935 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1937 /* depends on the current playing state of the pipeline. We might need to
1938 * queue this until we get EOS. */
1939 flags = GST_SEEK_FLAG_FLUSH;
1941 /* if range start was not supplied we must continue from current position.
1942 * but since we're doing a flushing seek, let us query the current position
1943 * so we end up at exactly the same position after the seek. */
1944 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
1945 if (current_position == -1) {
1946 GST_WARNING ("current position unknown");
1948 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
1949 GST_TIME_ARGS (current_position));
1950 start = current_position;
1951 start_type = GST_SEEK_TYPE_SET;
1952 flags |= GST_SEEK_FLAG_ACCURATE;
1955 /* only set keyframe flag when modifying start */
1956 if (start_type != GST_SEEK_TYPE_NONE)
1957 flags |= GST_SEEK_FLAG_KEY_UNIT;
1960 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
1961 GST_DEBUG ("not seeking because no position change");
1964 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
1966 media_streams_set_blocked (media, TRUE);
1968 /* FIXME, we only do forwards playback, no trick modes yet */
1969 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1970 flags, start_type, start, stop_type, stop);
1972 /* and block for the seek to complete */
1973 GST_INFO ("done seeking %d", res);
1977 g_rec_mutex_unlock (&priv->state_lock);
1979 /* wait until pipeline is prerolled again, this will also collect stats */
1980 if (!wait_preroll (media))
1981 goto preroll_failed;
1983 g_rec_mutex_lock (&priv->state_lock);
1984 GST_INFO ("prerolled again");
1987 GST_INFO ("no seek needed");
1990 g_rec_mutex_unlock (&priv->state_lock);
1997 g_rec_mutex_unlock (&priv->state_lock);
1998 GST_INFO ("media %p is not prepared", media);
2003 g_rec_mutex_unlock (&priv->state_lock);
2004 GST_INFO ("pipeline is not seekable");
2009 g_rec_mutex_unlock (&priv->state_lock);
2010 GST_WARNING ("conversion to npt not supported");
2015 g_rec_mutex_unlock (&priv->state_lock);
2016 GST_INFO ("seeking failed");
2017 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2022 GST_WARNING ("failed to preroll after seek");
2028 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2030 *blocked &= gst_rtsp_stream_is_blocking (stream);
2034 media_streams_blocking (GstRTSPMedia * media)
2036 gboolean blocking = TRUE;
2038 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2044 static GstStateChangeReturn
2045 set_state (GstRTSPMedia * media, GstState state)
2047 GstRTSPMediaPrivate *priv = media->priv;
2048 GstStateChangeReturn ret;
2050 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2052 ret = gst_element_set_state (priv->pipeline, state);
2057 static GstStateChangeReturn
2058 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2060 GstRTSPMediaPrivate *priv = media->priv;
2061 GstStateChangeReturn ret;
2063 GST_INFO ("set target state to %s for media %p",
2064 gst_element_state_get_name (state), media);
2065 priv->target_state = state;
2067 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2068 priv->target_state, NULL);
2071 ret = set_state (media, state);
2073 ret = GST_STATE_CHANGE_SUCCESS;
2078 /* called with state-lock */
2080 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2082 GstRTSPMediaPrivate *priv = media->priv;
2083 GstMessageType type;
2085 type = GST_MESSAGE_TYPE (message);
2088 case GST_MESSAGE_STATE_CHANGED:
2090 GstState old, new, pending;
2092 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2095 gst_message_parse_state_changed (message, &old, &new, &pending);
2097 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2098 gst_element_state_get_name (old), gst_element_state_get_name (new),
2099 gst_element_state_get_name (pending));
2100 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
2101 && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2102 GST_INFO ("%p: went to PAUSED, prepared now", media);
2103 collect_media_stats (media);
2105 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2106 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2111 case GST_MESSAGE_BUFFERING:
2115 gst_message_parse_buffering (message, &percent);
2117 /* no state management needed for live pipelines */
2121 if (percent == 100) {
2122 /* a 100% message means buffering is done */
2123 priv->buffering = FALSE;
2124 /* if the desired state is playing, go back */
2125 if (priv->target_state == GST_STATE_PLAYING) {
2126 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2127 set_state (media, GST_STATE_PLAYING);
2129 GST_INFO ("Buffering done");
2132 /* buffering busy */
2133 if (priv->buffering == FALSE) {
2134 if (priv->target_state == GST_STATE_PLAYING) {
2135 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2136 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2137 set_state (media, GST_STATE_PAUSED);
2139 GST_INFO ("Buffering ...");
2142 priv->buffering = TRUE;
2146 case GST_MESSAGE_LATENCY:
2148 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2151 case GST_MESSAGE_ERROR:
2156 gst_message_parse_error (message, &gerror, &debug);
2157 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2158 g_error_free (gerror);
2161 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2164 case GST_MESSAGE_WARNING:
2169 gst_message_parse_warning (message, &gerror, &debug);
2170 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2171 g_error_free (gerror);
2175 case GST_MESSAGE_ELEMENT:
2177 const GstStructure *s;
2179 s = gst_message_get_structure (message);
2180 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2181 GST_DEBUG ("media received blocking message");
2182 if (priv->blocked && media_streams_blocking (media)) {
2183 GST_DEBUG ("media is blocking");
2184 collect_media_stats (media);
2186 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2187 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2192 case GST_MESSAGE_STREAM_STATUS:
2194 case GST_MESSAGE_ASYNC_DONE:
2196 /* when we are dynamically adding pads, the addition of the udpsrc will
2197 * temporarily produce ASYNC_DONE messages. We have to ignore them and
2198 * wait for the final ASYNC_DONE after everything prerolled */
2199 GST_INFO ("%p: ignoring ASYNC_DONE", media);
2201 GST_INFO ("%p: got ASYNC_DONE", media);
2202 collect_media_stats (media);
2204 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2205 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2208 case GST_MESSAGE_EOS:
2209 GST_INFO ("%p: got EOS", media);
2211 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2212 GST_DEBUG ("shutting down after EOS");
2213 finish_unprepare (media);
2217 GST_INFO ("%p: got message type %d (%s)", media, type,
2218 gst_message_type_get_name (type));
2225 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2227 GstRTSPMediaPrivate *priv = media->priv;
2228 GstRTSPMediaClass *klass;
2231 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2233 g_rec_mutex_lock (&priv->state_lock);
2234 if (klass->handle_message)
2235 ret = klass->handle_message (media, message);
2238 g_rec_mutex_unlock (&priv->state_lock);
2244 watch_destroyed (GstRTSPMedia * media)
2246 GST_DEBUG_OBJECT (media, "source destroyed");
2247 g_object_unref (media);
2251 find_payload_element (GstElement * payloader)
2253 GstElement *pay = NULL;
2255 if (GST_IS_BIN (payloader)) {
2257 GValue item = { 0 };
2259 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2260 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2261 GstElement *element = (GstElement *) g_value_get_object (&item);
2262 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2266 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2270 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2271 pay = gst_object_ref (element);
2272 g_value_unset (&item);
2275 g_value_unset (&item);
2277 gst_iterator_free (iter);
2279 pay = g_object_ref (payloader);
2285 /* called from streaming threads */
2287 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2289 GstRTSPMediaPrivate *priv = media->priv;
2290 GstRTSPStream *stream;
2293 /* find the real payload element */
2294 pay = find_payload_element (element);
2295 stream = gst_rtsp_media_create_stream (media, pay, pad);
2296 gst_object_unref (pay);
2298 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2300 g_rec_mutex_lock (&priv->state_lock);
2301 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2304 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2306 /* we will be adding elements below that will cause ASYNC_DONE to be
2307 * posted in the bus. We want to ignore those messages until the
2308 * pipeline really prerolled. */
2309 priv->adding = TRUE;
2311 /* join the element in the PAUSED state because this callback is
2312 * called from the streaming thread and it is PAUSED */
2313 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2314 priv->rtpbin, GST_STATE_PAUSED)) {
2315 GST_WARNING ("failed to join bin element");
2318 priv->adding = FALSE;
2319 g_rec_mutex_unlock (&priv->state_lock);
2326 gst_rtsp_media_remove_stream (media, stream);
2327 g_rec_mutex_unlock (&priv->state_lock);
2328 GST_INFO ("ignore pad because we are not preparing");
2334 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2336 GstRTSPMediaPrivate *priv = media->priv;
2337 GstRTSPStream *stream;
2339 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2343 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2345 g_rec_mutex_lock (&priv->state_lock);
2346 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2347 g_rec_mutex_unlock (&priv->state_lock);
2349 gst_rtsp_media_remove_stream (media, stream);
2353 remove_fakesink (GstRTSPMediaPrivate * priv)
2355 GstElement *fakesink;
2357 g_mutex_lock (&priv->lock);
2358 if ((fakesink = priv->fakesink))
2359 gst_object_ref (fakesink);
2360 priv->fakesink = NULL;
2361 g_mutex_unlock (&priv->lock);
2364 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
2365 gst_element_set_state (fakesink, GST_STATE_NULL);
2366 gst_object_unref (fakesink);
2367 GST_INFO ("removed fakesink");
2372 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2374 GstRTSPMediaPrivate *priv = media->priv;
2376 GST_INFO ("no more pads");
2377 remove_fakesink (priv);
2380 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2382 struct _DynPaySignalHandlers
2384 gulong pad_added_handler;
2385 gulong pad_removed_handler;
2386 gulong no_more_pads_handler;
2390 start_preroll (GstRTSPMedia * media)
2392 GstRTSPMediaPrivate *priv = media->priv;
2393 GstStateChangeReturn ret;
2395 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2396 /* first go to PAUSED */
2397 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2400 case GST_STATE_CHANGE_SUCCESS:
2401 GST_INFO ("SUCCESS state change for media %p", media);
2402 priv->seekable = TRUE;
2404 case GST_STATE_CHANGE_ASYNC:
2405 GST_INFO ("ASYNC state change for media %p", media);
2406 priv->seekable = TRUE;
2408 case GST_STATE_CHANGE_NO_PREROLL:
2409 /* we need to go to PLAYING */
2410 GST_INFO ("NO_PREROLL state change: live media %p", media);
2411 /* FIXME we disable seeking for live streams for now. We should perform a
2412 * seeking query in preroll instead */
2413 priv->seekable = FALSE;
2414 priv->is_live = TRUE;
2415 if (!(priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
2416 /* start blocked to make sure nothing goes to the sink */
2417 media_streams_set_blocked (media, TRUE);
2419 ret = set_state (media, GST_STATE_PLAYING);
2420 if (ret == GST_STATE_CHANGE_FAILURE)
2423 case GST_STATE_CHANGE_FAILURE:
2431 GST_WARNING ("failed to preroll pipeline");
2437 wait_preroll (GstRTSPMedia * media)
2439 GstRTSPMediaStatus status;
2441 GST_DEBUG ("wait to preroll pipeline");
2443 /* wait until pipeline is prerolled */
2444 status = gst_rtsp_media_get_status (media);
2445 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2446 goto preroll_failed;
2452 GST_WARNING ("failed to preroll pipeline");
2458 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
2460 GstRTSPMediaPrivate *priv = media->priv;
2461 GstRTSPStream *stream = NULL;
2464 g_mutex_lock (&priv->lock);
2465 for (i = 0; i < priv->streams->len; i++) {
2466 stream = g_ptr_array_index (priv->streams, i);
2468 if (sessid == gst_rtsp_stream_get_index (stream))
2471 g_mutex_unlock (&priv->lock);
2473 return gst_rtsp_stream_request_aux_sender (stream, sessid);
2477 start_prepare (GstRTSPMedia * media)
2479 GstRTSPMediaPrivate *priv = media->priv;
2483 /* link streams we already have, other streams might appear when we have
2484 * dynamic elements */
2485 for (i = 0; i < priv->streams->len; i++) {
2486 GstRTSPStream *stream;
2488 stream = g_ptr_array_index (priv->streams, i);
2490 if (priv->rtx_time > 0) {
2491 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
2492 g_signal_connect (priv->rtpbin, "request-aux-sender",
2493 (GCallback) request_aux_sender, media);
2496 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2497 priv->rtpbin, GST_STATE_NULL)) {
2498 goto join_bin_failed;
2503 g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
2505 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2506 GstElement *elem = walk->data;
2507 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2509 GST_INFO ("adding callbacks for dynamic element %p", elem);
2511 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2512 (GCallback) pad_added_cb, media);
2513 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2514 (GCallback) pad_removed_cb, media);
2515 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2516 (GCallback) no_more_pads_cb, media);
2518 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2520 /* we add a fakesink here in order to make the state change async. We remove
2521 * the fakesink again in the no-more-pads callback. */
2522 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2523 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2526 if (!start_preroll (media))
2527 goto preroll_failed;
2533 GST_WARNING ("failed to join bin element");
2534 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2539 GST_WARNING ("failed to preroll pipeline");
2540 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2546 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2548 GstRTSPMediaPrivate *priv;
2549 GstRTSPMediaClass *klass;
2551 GMainContext *context;
2556 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2558 if (!klass->create_rtpbin)
2559 goto no_create_rtpbin;
2561 priv->rtpbin = klass->create_rtpbin (media);
2562 if (priv->rtpbin != NULL) {
2563 gboolean success = TRUE;
2565 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
2567 if (klass->setup_rtpbin)
2568 success = klass->setup_rtpbin (media, priv->rtpbin);
2570 if (success == FALSE) {
2571 gst_object_unref (priv->rtpbin);
2572 priv->rtpbin = NULL;
2575 if (priv->rtpbin == NULL)
2578 priv->thread = thread;
2579 context = (thread != NULL) ? (thread->context) : NULL;
2581 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2583 /* add the pipeline bus to our custom mainloop */
2584 priv->source = gst_bus_create_watch (bus);
2585 gst_object_unref (bus);
2587 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2588 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2590 priv->id = g_source_attach (priv->source, context);
2592 /* add stuff to the bin */
2593 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2595 /* do remainder in context */
2596 source = g_idle_source_new ();
2597 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
2598 g_source_attach (source, context);
2599 g_source_unref (source);
2606 GST_ERROR ("no create_rtpbin function");
2607 g_critical ("no create_rtpbin vmethod function set");
2612 GST_WARNING ("no rtpbin element");
2613 g_warning ("failed to create element 'rtpbin', check your installation");
2619 * gst_rtsp_media_prepare:
2620 * @media: a #GstRTSPMedia
2621 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2622 * bus handler or %NULL
2624 * Prepare @media for streaming. This function will create the objects
2625 * to manage the streaming. A pipeline must have been set on @media with
2626 * gst_rtsp_media_take_pipeline().
2628 * It will preroll the pipeline and collect vital information about the streams
2629 * such as the duration.
2631 * Returns: %TRUE on success.
2634 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2636 GstRTSPMediaPrivate *priv;
2637 GstRTSPMediaClass *klass;
2639 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2643 g_rec_mutex_lock (&priv->state_lock);
2644 priv->prepare_count++;
2646 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2647 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2650 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2653 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2654 goto not_unprepared;
2656 if (!priv->reusable && priv->reused)
2659 GST_INFO ("preparing media %p", media);
2661 /* reset some variables */
2662 priv->is_live = FALSE;
2663 priv->seekable = FALSE;
2664 priv->buffering = FALSE;
2666 /* we're preparing now */
2667 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2669 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2670 if (klass->prepare) {
2671 if (!klass->prepare (media, thread))
2672 goto prepare_failed;
2676 g_rec_mutex_unlock (&priv->state_lock);
2678 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2679 * able to do this async so that we don't block the server thread. */
2680 if (!wait_preroll (media))
2681 goto preroll_failed;
2683 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2685 GST_INFO ("object %p is prerolled", media);
2692 /* we are not going to use the giving thread, so stop it. */
2694 gst_rtsp_thread_stop (thread);
2699 GST_LOG ("media %p was prepared", media);
2700 /* we are not going to use the giving thread, so stop it. */
2702 gst_rtsp_thread_stop (thread);
2703 g_rec_mutex_unlock (&priv->state_lock);
2709 /* we are not going to use the giving thread, so stop it. */
2711 gst_rtsp_thread_stop (thread);
2712 GST_WARNING ("media %p was not unprepared", media);
2713 priv->prepare_count--;
2714 g_rec_mutex_unlock (&priv->state_lock);
2719 /* we are not going to use the giving thread, so stop it. */
2721 gst_rtsp_thread_stop (thread);
2722 priv->prepare_count--;
2723 g_rec_mutex_unlock (&priv->state_lock);
2724 GST_WARNING ("can not reuse media %p", media);
2729 /* we are not going to use the giving thread, so stop it. */
2731 gst_rtsp_thread_stop (thread);
2732 priv->prepare_count--;
2733 g_rec_mutex_unlock (&priv->state_lock);
2734 GST_ERROR ("failed to prepare media");
2739 GST_WARNING ("failed to preroll pipeline");
2740 gst_rtsp_media_unprepare (media);
2745 /* must be called with state-lock */
2747 finish_unprepare (GstRTSPMedia * media)
2749 GstRTSPMediaPrivate *priv = media->priv;
2753 GST_DEBUG ("shutting down");
2755 /* release the lock on shutdown, otherwise pad_added_cb might try to
2756 * acquire the lock and then we deadlock */
2757 g_rec_mutex_unlock (&priv->state_lock);
2758 set_state (media, GST_STATE_NULL);
2759 g_rec_mutex_lock (&priv->state_lock);
2760 remove_fakesink (priv);
2762 for (i = 0; i < priv->streams->len; i++) {
2763 GstRTSPStream *stream;
2765 GST_INFO ("Removing elements of stream %d from pipeline", i);
2767 stream = g_ptr_array_index (priv->streams, i);
2769 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2772 /* remove the pad signal handlers */
2773 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2774 GstElement *elem = walk->data;
2775 DynPaySignalHandlers *handlers;
2778 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2779 g_assert (handlers != NULL);
2781 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2782 g_signal_handler_disconnect (G_OBJECT (elem),
2783 handlers->pad_removed_handler);
2784 g_signal_handler_disconnect (G_OBJECT (elem),
2785 handlers->no_more_pads_handler);
2787 g_slice_free (DynPaySignalHandlers, handlers);
2790 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2791 priv->rtpbin = NULL;
2794 gst_object_unref (priv->nettime);
2795 priv->nettime = NULL;
2797 priv->reused = TRUE;
2798 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
2800 /* when the media is not reusable, this will effectively unref the media and
2802 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2804 /* the source has the last ref to the media */
2806 GST_DEBUG ("destroy source");
2807 g_source_destroy (priv->source);
2808 g_source_unref (priv->source);
2811 GST_DEBUG ("stop thread");
2812 gst_rtsp_thread_stop (priv->thread);
2816 /* called with state-lock */
2818 default_unprepare (GstRTSPMedia * media)
2820 GstRTSPMediaPrivate *priv = media->priv;
2822 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
2824 if (priv->eos_shutdown) {
2825 GST_DEBUG ("sending EOS for shutdown");
2826 /* ref so that we don't disappear */
2827 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2828 /* we need to go to playing again for the EOS to propagate, normally in this
2829 * state, nothing is receiving data from us anymore so this is ok. */
2830 set_state (media, GST_STATE_PLAYING);
2832 finish_unprepare (media);
2838 * gst_rtsp_media_unprepare:
2839 * @media: a #GstRTSPMedia
2841 * Unprepare @media. After this call, the media should be prepared again before
2842 * it can be used again. If the media is set to be non-reusable, a new instance
2845 * Returns: %TRUE on success.
2848 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2850 GstRTSPMediaPrivate *priv;
2853 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2857 g_rec_mutex_lock (&priv->state_lock);
2858 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2859 goto was_unprepared;
2861 priv->prepare_count--;
2862 if (priv->prepare_count > 0)
2865 GST_INFO ("unprepare media %p", media);
2867 media_streams_set_blocked (media, FALSE);
2868 set_target_state (media, GST_STATE_NULL, FALSE);
2871 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2872 GstRTSPMediaClass *klass;
2874 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2875 if (klass->unprepare)
2876 success = klass->unprepare (media);
2878 finish_unprepare (media);
2880 g_rec_mutex_unlock (&priv->state_lock);
2886 g_rec_mutex_unlock (&priv->state_lock);
2887 GST_INFO ("media %p was already unprepared", media);
2892 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2893 g_rec_mutex_unlock (&priv->state_lock);
2898 /* should be called with state-lock */
2900 get_clock_unlocked (GstRTSPMedia * media)
2902 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2903 GST_DEBUG_OBJECT (media, "media was not prepared");
2906 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2910 * gst_rtsp_media_get_clock:
2911 * @media: a #GstRTSPMedia
2913 * Get the clock that is used by the pipeline in @media.
2915 * @media must be prepared before this method returns a valid clock object.
2917 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2920 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2923 GstRTSPMediaPrivate *priv;
2925 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2929 g_rec_mutex_lock (&priv->state_lock);
2930 clock = get_clock_unlocked (media);
2931 g_rec_mutex_unlock (&priv->state_lock);
2937 * gst_rtsp_media_get_base_time:
2938 * @media: a #GstRTSPMedia
2940 * Get the base_time that is used by the pipeline in @media.
2942 * @media must be prepared before this method returns a valid base_time.
2944 * Returns: the base_time used by @media.
2947 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2949 GstClockTime result;
2950 GstRTSPMediaPrivate *priv;
2952 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2956 g_rec_mutex_lock (&priv->state_lock);
2957 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2960 result = gst_element_get_base_time (media->priv->pipeline);
2961 g_rec_mutex_unlock (&priv->state_lock);
2968 g_rec_mutex_unlock (&priv->state_lock);
2969 GST_DEBUG_OBJECT (media, "media was not prepared");
2970 return GST_CLOCK_TIME_NONE;
2975 * gst_rtsp_media_get_time_provider:
2976 * @media: a #GstRTSPMedia
2977 * @address: (allow-none): an address or %NULL
2978 * @port: a port or 0
2980 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
2981 * will listen on @address and @port for client time requests.
2983 * Returns: (transfer full): the #GstNetTimeProvider of @media.
2985 GstNetTimeProvider *
2986 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
2989 GstRTSPMediaPrivate *priv;
2990 GstNetTimeProvider *provider = NULL;
2992 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2996 g_rec_mutex_lock (&priv->state_lock);
2997 if (priv->time_provider) {
2998 if ((provider = priv->nettime) == NULL) {
3001 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3002 provider = gst_net_time_provider_new (clock, address, port);
3003 gst_object_unref (clock);
3005 priv->nettime = provider;
3009 g_rec_mutex_unlock (&priv->state_lock);
3012 gst_object_ref (provider);
3018 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3020 return gst_rtsp_sdp_from_media (sdp, info, media);
3024 * gst_rtsp_media_setup_sdp:
3025 * @media: a #GstRTSPMedia
3026 * @sdp: (transfer none): a #GstSDPMessage
3027 * @info: (transfer none): a #GstSDPInfo
3029 * Add @media specific info to @sdp. @info is used to configure the connection
3030 * information in the SDP.
3032 * Returns: TRUE on success.
3035 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3038 GstRTSPMediaPrivate *priv;
3039 GstRTSPMediaClass *klass;
3042 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3043 g_return_val_if_fail (sdp != NULL, FALSE);
3044 g_return_val_if_fail (info != NULL, FALSE);
3048 g_rec_mutex_lock (&priv->state_lock);
3050 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3052 if (!klass->setup_sdp)
3055 res = klass->setup_sdp (media, sdp, info);
3057 g_rec_mutex_unlock (&priv->state_lock);
3064 g_rec_mutex_unlock (&priv->state_lock);
3065 GST_ERROR ("no setup_sdp function");
3066 g_critical ("no setup_sdp vmethod function set");
3071 static const gchar *
3072 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
3081 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
3084 if (sscanf (attr, "%d ", &val) != 1)
3093 #define PARSE_INT(p, del, res) \
3096 p = strstr (p, del); \
3106 #define PARSE_STRING(p, del, res) \
3109 p = strstr (p, del); \
3121 #define SKIP_SPACES(p) \
3122 while (*p && g_ascii_isspace (*p)) \
3127 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
3130 parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
3131 gint * rate, gchar ** params)
3135 p = (gchar *) rtpmap;
3137 PARSE_INT (p, " ", *payload);
3145 PARSE_STRING (p, "/", *name);
3146 if (*name == NULL) {
3147 GST_DEBUG ("no rate, name %s", p);
3148 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
3149 * streams seem to omit the rate. */
3156 p = strstr (p, "/");
3174 * Mapping of caps to and from SDP fields:
3176 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
3177 * a=framesize:<payload> <width>-<height>
3178 * a=fmtp:<payload> <param>[=<value>];...
3181 media_to_caps (gint pt, const GstSDPMedia * media)
3184 const gchar *rtpmap;
3186 const gchar *framesize;
3189 gchar *params = NULL;
3195 /* get and parse rtpmap */
3196 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
3199 ret = parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
3201 g_warning ("error parsing rtpmap, ignoring");
3205 /* dynamic payloads need rtpmap or we fail */
3206 if (rtpmap == NULL && pt >= 96)
3209 /* check if we have a rate, if not, we need to look up the rate from the
3210 * default rates based on the payload types. */
3212 const GstRTPPayloadInfo *info;
3214 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
3215 /* dynamic types, use media and encoding_name */
3216 tmp = g_ascii_strdown (media->media, -1);
3217 info = gst_rtp_payload_info_for_name (tmp, name);
3220 /* static types, use payload type */
3221 info = gst_rtp_payload_info_for_pt (pt);
3225 if ((rate = info->clock_rate) == 0)
3228 /* we fail if we cannot find one */
3233 tmp = g_ascii_strdown (media->media, -1);
3234 caps = gst_caps_new_simple ("application/x-unknown",
3235 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
3237 s = gst_caps_get_structure (caps, 0);
3239 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
3241 /* encoding name must be upper case */
3243 tmp = g_ascii_strup (name, -1);
3244 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
3248 /* params must be lower case */
3249 if (params != NULL) {
3250 tmp = g_ascii_strdown (params, -1);
3251 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
3255 /* parse optional fmtp: field */
3256 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
3262 /* p is now of the format <payload> <param>[=<value>];... */
3263 PARSE_INT (p, " ", payload);
3264 if (payload != -1 && payload == pt) {
3268 /* <param>[=<value>] are separated with ';' */
3269 pairs = g_strsplit (p, ";", 0);
3270 for (i = 0; pairs[i]; i++) {
3272 const gchar *val, *key;
3274 const gchar *reserved_keys[] =
3275 { "media", "payload", "clock-rate", "encoding-name",
3279 /* the key may not have a '=', the value can have other '='s */
3280 valpos = strstr (pairs[i], "=");
3282 /* we have a '=' and thus a value, remove the '=' with \0 */
3284 /* value is everything between '=' and ';'. We split the pairs at ;
3285 * boundaries so we can take the remainder of the value. Some servers
3286 * put spaces around the value which we strip off here. Alternatively
3287 * we could strip those spaces in the depayloaders should these spaces
3288 * actually carry any meaning in the future. */
3289 val = g_strstrip (valpos + 1);
3291 /* simple <param>;.. is translated into <param>=1;... */
3294 /* strip the key of spaces, convert key to lowercase but not the value. */
3295 key = g_strstrip (pairs[i]);
3297 /* skip keys from the fmtp, which we already use ourselves for the
3298 * caps. Some software is adding random things like clock-rate into
3299 * the fmtp, and we would otherwise here set a string-typed clock-rate
3300 * in the caps... and thus fail to create valid RTP caps
3302 for (j = 0; j < G_N_ELEMENTS (reserved_keys); j++) {
3303 if (g_ascii_strcasecmp (reserved_keys[i], key) == 0) {
3309 if (strlen (key) > 1) {
3310 tmp = g_ascii_strdown (key, -1);
3311 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
3319 /* parse framesize: field */
3320 if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
3323 /* p is now of the format <payload> <width>-<height> */
3324 p = (gchar *) framesize;
3326 PARSE_INT (p, " ", payload);
3327 if (payload != -1 && payload == pt) {
3328 gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
3336 g_warning ("rtpmap type not given for dynamic payload %d", pt);
3341 g_warning ("rate unknown for payload type %d", pt);
3347 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
3349 gboolean res = FALSE;
3353 GstMIKEYMessage *msg;
3354 const GstMIKEYPayload *payload;
3355 const gchar *srtp_cipher;
3356 const gchar *srtp_auth;
3358 p = (gchar *) keymgmt;
3364 PARSE_STRING (p, " ", kmpid);
3365 if (!g_str_equal (kmpid, "mikey"))
3368 data = g_base64_decode (p, &size);
3372 msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
3377 srtp_cipher = "aes-128-icm";
3378 srtp_auth = "hmac-sha1-80";
3380 /* check the Security policy if any */
3381 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
3382 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
3385 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
3388 len = gst_mikey_payload_sp_get_n_params (payload);
3389 for (i = 0; i < len; i++) {
3390 const GstMIKEYPayloadSPParam *param =
3391 gst_mikey_payload_sp_get_param (payload, i);
3393 switch (param->type) {
3394 case GST_MIKEY_SP_SRTP_ENC_ALG:
3395 switch (param->val[0]) {
3397 srtp_cipher = "null";
3401 srtp_cipher = "aes-128-icm";
3407 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
3408 switch (param->val[0]) {
3409 case AES_128_KEY_LEN:
3410 srtp_cipher = "aes-128-icm";
3412 case AES_256_KEY_LEN:
3413 srtp_cipher = "aes-256-icm";
3419 case GST_MIKEY_SP_SRTP_AUTH_ALG:
3420 switch (param->val[0]) {
3426 srtp_auth = "hmac-sha1-80";
3432 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
3433 switch (param->val[0]) {
3434 case HMAC_32_KEY_LEN:
3435 srtp_auth = "hmac-sha1-32";
3437 case HMAC_80_KEY_LEN:
3438 srtp_auth = "hmac-sha1-80";
3444 case GST_MIKEY_SP_SRTP_SRTP_ENC:
3446 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
3454 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
3457 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
3458 const GstMIKEYPayload *sub;
3459 GstMIKEYPayloadKeyData *pkd;
3462 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
3465 if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
3468 if (sub->type != GST_MIKEY_PT_KEY_DATA)
3471 pkd = (GstMIKEYPayloadKeyData *) sub;
3473 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
3475 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
3478 gst_caps_set_simple (caps,
3479 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
3480 "srtp-auth", G_TYPE_STRING, srtp_auth,
3481 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
3482 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
3486 gst_mikey_message_unref (msg);
3492 * Mapping SDP attributes to caps
3494 * prepend 'a-' to IANA registered sdp attributes names
3495 * (ie: not prefixed with 'x-') in order to avoid
3496 * collision with gstreamer standard caps properties names
3499 sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
3501 if (attributes->len > 0) {
3505 s = gst_caps_get_structure (caps, 0);
3507 for (i = 0; i < attributes->len; i++) {
3508 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
3509 gchar *tofree, *key;
3513 /* skip some of the attribute we already handle */
3514 if (!strcmp (key, "fmtp"))
3516 if (!strcmp (key, "rtpmap"))
3518 if (!strcmp (key, "control"))
3520 if (!strcmp (key, "range"))
3522 if (!strcmp (key, "framesize"))
3524 if (g_str_equal (key, "key-mgmt")) {
3525 parse_keymgmt (attr->value, caps);
3529 /* string must be valid UTF8 */
3530 if (!g_utf8_validate (attr->value, -1, NULL))
3533 if (!g_str_has_prefix (key, "x-"))
3534 tofree = key = g_strdup_printf ("a-%s", key);
3538 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
3539 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
3546 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3548 GstRTSPMediaPrivate *priv = media->priv;
3551 medias_len = gst_sdp_message_medias_len (sdp);
3552 if (medias_len != priv->streams->len) {
3553 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3554 priv->streams->len, medias_len);
3558 for (i = 0; i < medias_len; i++) {
3559 const gchar *proto, *media_type;
3560 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3561 GstRTSPStream *stream;
3562 gint j, formats_len;
3563 const gchar *control;
3564 GstRTSPProfile profile, profiles;
3566 stream = g_ptr_array_index (priv->streams, i);
3568 /* TODO: Should we do something with the other SDP information? */
3571 proto = gst_sdp_media_get_proto (sdp_media);
3572 if (proto == NULL) {
3573 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3577 if (g_str_equal (proto, "RTP/AVP")) {
3578 media_type = "application/x-rtp";
3579 profile = GST_RTSP_PROFILE_AVP;
3580 } else if (g_str_equal (proto, "RTP/SAVP")) {
3581 media_type = "application/x-srtp";
3582 profile = GST_RTSP_PROFILE_SAVP;
3583 } else if (g_str_equal (proto, "RTP/AVPF")) {
3584 media_type = "application/x-rtp";
3585 profile = GST_RTSP_PROFILE_AVPF;
3586 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3587 media_type = "application/x-srtp";
3588 profile = GST_RTSP_PROFILE_SAVPF;
3590 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3594 profiles = gst_rtsp_stream_get_profiles (stream);
3595 if ((profiles & profile) == 0) {
3596 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3600 formats_len = gst_sdp_media_formats_len (sdp_media);
3601 for (j = 0; j < formats_len; j++) {
3606 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3608 GST_DEBUG (" looking at %d pt: %d", j, pt);
3611 caps = media_to_caps (pt, sdp_media);
3613 GST_WARNING (" skipping pt %d without caps", pt);
3617 /* do some tweaks */
3618 GST_DEBUG ("mapping sdp session level attributes to caps");
3619 sdp_attributes_to_caps (sdp->attributes, caps);
3620 GST_DEBUG ("mapping sdp media level attributes to caps");
3621 sdp_attributes_to_caps (sdp_media->attributes, caps);
3623 s = gst_caps_get_structure (caps, 0);
3624 gst_structure_set_name (s, media_type);
3626 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3627 gst_caps_unref (caps);
3630 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3632 gst_rtsp_stream_set_control (stream, control);
3640 * gst_rtsp_media_handle_sdp:
3641 * @media: a #GstRTSPMedia
3642 * @sdp: (transfer none): a #GstSDPMessage
3644 * Configure an SDP on @media for receiving streams
3646 * Returns: TRUE on success.
3649 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3651 GstRTSPMediaPrivate *priv;
3652 GstRTSPMediaClass *klass;
3655 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3656 g_return_val_if_fail (sdp != NULL, FALSE);
3660 g_rec_mutex_lock (&priv->state_lock);
3662 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3664 if (!klass->handle_sdp)
3667 res = klass->handle_sdp (media, sdp);
3669 g_rec_mutex_unlock (&priv->state_lock);
3676 g_rec_mutex_unlock (&priv->state_lock);
3677 GST_ERROR ("no handle_sdp function");
3678 g_critical ("no handle_sdp vmethod function set");
3684 do_set_seqnum (GstRTSPStream * stream)
3687 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3688 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3691 /* call with state_lock */
3693 default_suspend (GstRTSPMedia * media)
3695 GstRTSPMediaPrivate *priv = media->priv;
3696 GstStateChangeReturn ret;
3697 gboolean unblock = FALSE;
3699 switch (priv->suspend_mode) {
3700 case GST_RTSP_SUSPEND_MODE_NONE:
3701 GST_DEBUG ("media %p no suspend", media);
3703 case GST_RTSP_SUSPEND_MODE_PAUSE:
3704 GST_DEBUG ("media %p suspend to PAUSED", media);
3705 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3706 if (ret == GST_STATE_CHANGE_FAILURE)
3710 case GST_RTSP_SUSPEND_MODE_RESET:
3711 GST_DEBUG ("media %p suspend to NULL", media);
3712 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3713 if (ret == GST_STATE_CHANGE_FAILURE)
3715 /* Because payloader needs to set the sequence number as
3716 * monotonic, we need to preserve the sequence number
3717 * after pause. (otherwise going from pause to play, which
3718 * is actually from NULL to PLAY will create a new sequence
3720 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3727 /* let the streams do the state changes freely, if any */
3729 media_streams_set_blocked (media, FALSE);
3736 GST_WARNING ("failed changing pipeline's state for media %p", media);
3742 * gst_rtsp_media_suspend:
3743 * @media: a #GstRTSPMedia
3745 * Suspend @media. The state of the pipeline managed by @media is set to
3746 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3747 * with gst_rtsp_media_unsuspend()
3749 * @media must be prepared with gst_rtsp_media_prepare();
3751 * Returns: %TRUE on success.
3754 gst_rtsp_media_suspend (GstRTSPMedia * media)
3756 GstRTSPMediaPrivate *priv = media->priv;
3757 GstRTSPMediaClass *klass;
3759 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3761 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3763 g_rec_mutex_lock (&priv->state_lock);
3764 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3767 /* don't attempt to suspend when something is busy */
3768 if (priv->n_active > 0)
3771 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3772 if (klass->suspend) {
3773 if (!klass->suspend (media))
3774 goto suspend_failed;
3777 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3779 g_rec_mutex_unlock (&priv->state_lock);
3786 g_rec_mutex_unlock (&priv->state_lock);
3787 GST_WARNING ("media %p was not prepared", media);
3792 g_rec_mutex_unlock (&priv->state_lock);
3793 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3794 GST_WARNING ("failed to suspend media %p", media);
3799 /* call with state_lock */
3801 default_unsuspend (GstRTSPMedia * media)
3803 GstRTSPMediaPrivate *priv = media->priv;
3805 switch (priv->suspend_mode) {
3806 case GST_RTSP_SUSPEND_MODE_NONE:
3807 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3809 case GST_RTSP_SUSPEND_MODE_PAUSE:
3810 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3812 case GST_RTSP_SUSPEND_MODE_RESET:
3814 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3815 if (!start_preroll (media))
3817 g_rec_mutex_unlock (&priv->state_lock);
3819 if (!wait_preroll (media))
3820 goto preroll_failed;
3822 g_rec_mutex_lock (&priv->state_lock);
3833 GST_WARNING ("failed to preroll pipeline");
3838 GST_WARNING ("failed to preroll pipeline");
3844 * gst_rtsp_media_unsuspend:
3845 * @media: a #GstRTSPMedia
3847 * Unsuspend @media if it was in a suspended state. This method does nothing
3848 * when the media was not in the suspended state.
3850 * Returns: %TRUE on success.
3853 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
3855 GstRTSPMediaPrivate *priv = media->priv;
3856 GstRTSPMediaClass *klass;
3858 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3860 g_rec_mutex_lock (&priv->state_lock);
3861 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3864 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3865 if (klass->unsuspend) {
3866 if (!klass->unsuspend (media))
3867 goto unsuspend_failed;
3871 g_rec_mutex_unlock (&priv->state_lock);
3878 g_rec_mutex_unlock (&priv->state_lock);
3879 GST_WARNING ("failed to unsuspend media %p", media);
3880 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3885 /* must be called with state-lock */
3887 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
3889 GstRTSPMediaPrivate *priv = media->priv;
3891 if (state == GST_STATE_NULL) {
3892 gst_rtsp_media_unprepare (media);
3894 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
3895 set_target_state (media, state, FALSE);
3896 /* when we are buffering, don't update the state yet, this will be done
3897 * when buffering finishes */
3898 if (priv->buffering) {
3899 GST_INFO ("Buffering busy, delay state change");
3901 if (state == GST_STATE_PLAYING)
3902 /* make sure pads are not blocking anymore when going to PLAYING */
3903 media_streams_set_blocked (media, FALSE);
3905 set_state (media, state);
3907 /* and suspend after pause */
3908 if (state == GST_STATE_PAUSED)
3909 gst_rtsp_media_suspend (media);
3915 * gst_rtsp_media_set_pipeline_state:
3916 * @media: a #GstRTSPMedia
3917 * @state: the target state of the pipeline
3919 * Set the state of the pipeline managed by @media to @state
3922 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
3924 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
3926 g_rec_mutex_lock (&media->priv->state_lock);
3927 media_set_pipeline_state_locked (media, state);
3928 g_rec_mutex_unlock (&media->priv->state_lock);
3932 * gst_rtsp_media_set_state:
3933 * @media: a #GstRTSPMedia
3934 * @state: the target state of the media
3935 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
3936 * a #GPtrArray of #GstRTSPStreamTransport pointers
3938 * Set the state of @media to @state and for the transports in @transports.
3940 * @media must be prepared with gst_rtsp_media_prepare();
3942 * Returns: %TRUE on success.
3945 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
3946 GPtrArray * transports)
3948 GstRTSPMediaPrivate *priv;
3950 gboolean activate, deactivate, do_state;
3953 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3954 g_return_val_if_fail (transports != NULL, FALSE);
3958 g_rec_mutex_lock (&priv->state_lock);
3959 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
3961 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
3962 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3965 /* NULL and READY are the same */
3966 if (state == GST_STATE_READY)
3967 state = GST_STATE_NULL;
3969 activate = deactivate = FALSE;
3971 GST_INFO ("going to state %s media %p, target state %s",
3972 gst_element_state_get_name (state), media,
3973 gst_element_state_get_name (priv->target_state));
3976 case GST_STATE_NULL:
3977 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
3978 if (priv->target_state >= GST_STATE_PAUSED)
3981 case GST_STATE_PAUSED:
3982 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
3983 if (priv->target_state == GST_STATE_PLAYING)
3986 case GST_STATE_PLAYING:
3987 /* we're going to PLAYING, activate */
3993 old_active = priv->n_active;
3995 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
3996 activate, deactivate);
3997 for (i = 0; i < transports->len; i++) {
3998 GstRTSPStreamTransport *trans;
4000 /* we need a non-NULL entry in the array */
4001 trans = g_ptr_array_index (transports, i);
4006 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4008 } else if (deactivate) {
4009 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4014 /* we just activated the first media, do the playing state change */
4015 if (old_active == 0 && activate)
4017 /* if we have no more active media, do the downward state changes */
4018 else if (priv->n_active == 0)
4023 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4026 if (priv->target_state != state) {
4028 media_set_pipeline_state_locked (media, state);
4030 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4034 /* remember where we are */
4035 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4036 old_active != priv->n_active))
4037 collect_media_stats (media);
4039 g_rec_mutex_unlock (&priv->state_lock);
4046 GST_WARNING ("media %p was not prepared", media);
4047 g_rec_mutex_unlock (&priv->state_lock);
4052 GST_WARNING ("media %p in error status while changing to state %d",
4054 if (state == GST_STATE_NULL) {
4055 for (i = 0; i < transports->len; i++) {
4056 GstRTSPStreamTransport *trans;
4058 /* we need a non-NULL entry in the array */
4059 trans = g_ptr_array_index (transports, i);
4063 gst_rtsp_stream_transport_set_active (trans, FALSE);
4067 g_rec_mutex_unlock (&priv->state_lock);
4073 * gst_rtsp_media_set_transport_mode:
4074 * @media: a #GstRTSPMedia
4075 * @mode: the new value
4077 * Sets if the media pipeline can work in PLAY or RECORD mode
4080 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4081 GstRTSPTransportMode mode)
4083 GstRTSPMediaPrivate *priv;
4085 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4089 g_mutex_lock (&priv->lock);
4090 priv->transport_mode = mode;
4091 g_mutex_unlock (&priv->lock);
4095 * gst_rtsp_media_get_transport_mode:
4096 * @media: a #GstRTSPMedia
4098 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4100 * Returns: The transport mode.
4102 GstRTSPTransportMode
4103 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4105 GstRTSPMediaPrivate *priv;
4106 GstRTSPTransportMode res;
4108 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4112 g_mutex_lock (&priv->lock);
4113 res = priv->transport_mode;
4114 g_mutex_unlock (&priv->lock);