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_start_preroll (GstRTSPMedia * media);
200 static gboolean default_unprepare (GstRTSPMedia * media);
201 static gboolean default_suspend (GstRTSPMedia * media);
202 static gboolean default_unsuspend (GstRTSPMedia * media);
203 static gboolean default_convert_range (GstRTSPMedia * media,
204 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
205 static gboolean default_query_position (GstRTSPMedia * media,
207 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
208 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
209 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
211 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
212 static gboolean default_start_prepare (GstRTSPMedia * media);
214 static gboolean wait_preroll (GstRTSPMedia * media);
216 static GstElement *find_payload_element (GstElement * payloader);
218 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
220 #define C_ENUM(v) ((gint) v)
223 gst_rtsp_suspend_mode_get_type (void)
226 static const GEnumValue values[] = {
227 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
228 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
230 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
235 if (g_once_init_enter (&id)) {
236 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
237 g_once_init_leave (&id, tmp);
242 #define C_FLAGS(v) ((guint) v)
245 gst_rtsp_transport_mode_get_type (void)
248 static const GFlagsValue values[] = {
249 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
251 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
256 if (g_once_init_enter (&id)) {
257 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
258 g_once_init_leave (&id, tmp);
263 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
266 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
268 GObjectClass *gobject_class;
270 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
272 gobject_class = G_OBJECT_CLASS (klass);
274 gobject_class->get_property = gst_rtsp_media_get_property;
275 gobject_class->set_property = gst_rtsp_media_set_property;
276 gobject_class->finalize = gst_rtsp_media_finalize;
278 g_object_class_install_property (gobject_class, PROP_SHARED,
279 g_param_spec_boolean ("shared", "Shared",
280 "If this media pipeline can be shared", DEFAULT_SHARED,
281 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
283 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
284 g_param_spec_enum ("suspend-mode", "Suspend Mode",
285 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
286 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
288 g_object_class_install_property (gobject_class, PROP_REUSABLE,
289 g_param_spec_boolean ("reusable", "Reusable",
290 "If this media pipeline can be reused after an unprepare",
291 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
293 g_object_class_install_property (gobject_class, PROP_PROFILES,
294 g_param_spec_flags ("profiles", "Profiles",
295 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
296 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
298 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
299 g_param_spec_flags ("protocols", "Protocols",
300 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
301 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
303 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
304 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
305 "Send an EOS event to the pipeline before unpreparing",
306 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
308 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
309 g_param_spec_uint ("buffer-size", "Buffer Size",
310 "The kernel UDP buffer size to use", 0, G_MAXUINT,
311 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
313 g_object_class_install_property (gobject_class, PROP_ELEMENT,
314 g_param_spec_object ("element", "The Element",
315 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
316 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
318 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
319 g_param_spec_boolean ("time-provider", "Time Provider",
320 "Use a NetTimeProvider for clients",
321 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
323 g_object_class_install_property (gobject_class, PROP_LATENCY,
324 g_param_spec_uint ("latency", "Latency",
325 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
326 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
328 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
329 g_param_spec_flags ("transport-mode", "Transport Mode",
330 "If this media pipeline can be used for PLAY or RECORD",
331 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
332 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
334 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
335 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
336 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
337 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
339 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
340 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
341 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
342 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
343 GST_TYPE_RTSP_STREAM);
345 gst_rtsp_media_signals[SIGNAL_PREPARED] =
346 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
347 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
348 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
350 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
351 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
352 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
353 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
355 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
356 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
357 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
358 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
360 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
361 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
362 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
363 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
365 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
367 klass->handle_message = default_handle_message;
368 klass->prepare = default_prepare;
369 klass->start_preroll = default_start_preroll;
370 klass->unprepare = default_unprepare;
371 klass->suspend = default_suspend;
372 klass->unsuspend = default_unsuspend;
373 klass->convert_range = default_convert_range;
374 klass->query_position = default_query_position;
375 klass->query_stop = default_query_stop;
376 klass->create_rtpbin = default_create_rtpbin;
377 klass->setup_sdp = default_setup_sdp;
378 klass->handle_sdp = default_handle_sdp;
379 klass->start_prepare = default_start_prepare;
383 gst_rtsp_media_init (GstRTSPMedia * media)
385 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
389 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
390 g_mutex_init (&priv->lock);
391 g_cond_init (&priv->cond);
392 g_rec_mutex_init (&priv->state_lock);
394 priv->shared = DEFAULT_SHARED;
395 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
396 priv->reusable = DEFAULT_REUSABLE;
397 priv->profiles = DEFAULT_PROFILES;
398 priv->protocols = DEFAULT_PROTOCOLS;
399 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
400 priv->buffer_size = DEFAULT_BUFFER_SIZE;
401 priv->time_provider = DEFAULT_TIME_PROVIDER;
402 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
406 gst_rtsp_media_finalize (GObject * obj)
408 GstRTSPMediaPrivate *priv;
411 media = GST_RTSP_MEDIA (obj);
414 GST_INFO ("finalize media %p", media);
416 if (priv->permissions)
417 gst_rtsp_permissions_unref (priv->permissions);
419 g_ptr_array_unref (priv->streams);
421 g_list_free_full (priv->dynamic, gst_object_unref);
424 gst_object_unref (priv->pipeline);
426 gst_object_unref (priv->nettime);
427 gst_object_unref (priv->element);
429 g_object_unref (priv->pool);
431 g_list_free (priv->payloads);
432 g_mutex_clear (&priv->lock);
433 g_cond_clear (&priv->cond);
434 g_rec_mutex_clear (&priv->state_lock);
436 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
440 gst_rtsp_media_get_property (GObject * object, guint propid,
441 GValue * value, GParamSpec * pspec)
443 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
447 g_value_set_object (value, media->priv->element);
450 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
452 case PROP_SUSPEND_MODE:
453 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
456 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
459 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
462 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
464 case PROP_EOS_SHUTDOWN:
465 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
467 case PROP_BUFFER_SIZE:
468 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
470 case PROP_TIME_PROVIDER:
471 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
474 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
476 case PROP_TRANSPORT_MODE:
477 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
480 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
485 gst_rtsp_media_set_property (GObject * object, guint propid,
486 const GValue * value, GParamSpec * pspec)
488 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
492 media->priv->element = g_value_get_object (value);
493 gst_object_ref_sink (media->priv->element);
496 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
498 case PROP_SUSPEND_MODE:
499 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
502 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
505 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
508 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
510 case PROP_EOS_SHUTDOWN:
511 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
513 case PROP_BUFFER_SIZE:
514 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
516 case PROP_TIME_PROVIDER:
517 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
520 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
522 case PROP_TRANSPORT_MODE:
523 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
526 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
534 } DoQueryPositionData;
537 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
541 if (gst_rtsp_stream_query_position (stream, &tmp)) {
542 data->position = MAX (data->position, tmp);
548 default_query_position (GstRTSPMedia * media, gint64 * position)
550 GstRTSPMediaPrivate *priv;
551 DoQueryPositionData data;
558 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
560 *position = data.position;
572 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
576 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
577 data->stop = MAX (data->stop, tmp);
583 default_query_stop (GstRTSPMedia * media, gint64 * stop)
585 GstRTSPMediaPrivate *priv;
586 DoQueryStopData data;
593 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
601 default_create_rtpbin (GstRTSPMedia * media)
605 rtpbin = gst_element_factory_make ("rtpbin", NULL);
610 /* must be called with state lock */
612 collect_media_stats (GstRTSPMedia * media)
614 GstRTSPMediaPrivate *priv = media->priv;
615 gint64 position = 0, stop = -1;
617 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
618 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
621 priv->range.unit = GST_RTSP_RANGE_NPT;
623 GST_INFO ("collect media stats");
626 priv->range.min.type = GST_RTSP_TIME_NOW;
627 priv->range.min.seconds = -1;
628 priv->range_start = -1;
629 priv->range.max.type = GST_RTSP_TIME_END;
630 priv->range.max.seconds = -1;
631 priv->range_stop = -1;
633 GstRTSPMediaClass *klass;
636 klass = GST_RTSP_MEDIA_GET_CLASS (media);
638 /* get the position */
640 if (klass->query_position)
641 ret = klass->query_position (media, &position);
644 GST_INFO ("position query failed");
648 /* get the current segment stop */
650 if (klass->query_stop)
651 ret = klass->query_stop (media, &stop);
654 GST_INFO ("stop query failed");
658 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
659 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
661 if (position == -1) {
662 priv->range.min.type = GST_RTSP_TIME_NOW;
663 priv->range.min.seconds = -1;
664 priv->range_start = -1;
666 priv->range.min.type = GST_RTSP_TIME_SECONDS;
667 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
668 priv->range_start = position;
671 priv->range.max.type = GST_RTSP_TIME_END;
672 priv->range.max.seconds = -1;
673 priv->range_stop = -1;
675 priv->range.max.type = GST_RTSP_TIME_SECONDS;
676 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
677 priv->range_stop = stop;
683 * gst_rtsp_media_new:
684 * @element: (transfer full): a #GstElement
686 * Create a new #GstRTSPMedia instance. @element is the bin element that
687 * provides the different streams. The #GstRTSPMedia object contains the
688 * element to produce RTP data for one or more related (audio/video/..)
691 * Ownership is taken of @element.
693 * Returns: (transfer full): a new #GstRTSPMedia object.
696 gst_rtsp_media_new (GstElement * element)
698 GstRTSPMedia *result;
700 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
702 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
708 * gst_rtsp_media_get_element:
709 * @media: a #GstRTSPMedia
711 * Get the element that was used when constructing @media.
713 * Returns: (transfer full): a #GstElement. Unref after usage.
716 gst_rtsp_media_get_element (GstRTSPMedia * media)
718 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
720 return gst_object_ref (media->priv->element);
724 * gst_rtsp_media_take_pipeline:
725 * @media: a #GstRTSPMedia
726 * @pipeline: (transfer full): a #GstPipeline
728 * Set @pipeline as the #GstPipeline for @media. Ownership is
729 * taken of @pipeline.
732 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
734 GstRTSPMediaPrivate *priv;
736 GstNetTimeProvider *nettime;
738 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
739 g_return_if_fail (GST_IS_PIPELINE (pipeline));
743 g_mutex_lock (&priv->lock);
744 old = priv->pipeline;
745 priv->pipeline = GST_ELEMENT_CAST (pipeline);
746 nettime = priv->nettime;
747 priv->nettime = NULL;
748 g_mutex_unlock (&priv->lock);
751 gst_object_unref (old);
754 gst_object_unref (nettime);
756 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
760 * gst_rtsp_media_set_permissions:
761 * @media: a #GstRTSPMedia
762 * @permissions: (transfer none): a #GstRTSPPermissions
764 * Set @permissions on @media.
767 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
768 GstRTSPPermissions * permissions)
770 GstRTSPMediaPrivate *priv;
772 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
776 g_mutex_lock (&priv->lock);
777 if (priv->permissions)
778 gst_rtsp_permissions_unref (priv->permissions);
779 if ((priv->permissions = permissions))
780 gst_rtsp_permissions_ref (permissions);
781 g_mutex_unlock (&priv->lock);
785 * gst_rtsp_media_get_permissions:
786 * @media: a #GstRTSPMedia
788 * Get the permissions object from @media.
790 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
793 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
795 GstRTSPMediaPrivate *priv;
796 GstRTSPPermissions *result;
798 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
802 g_mutex_lock (&priv->lock);
803 if ((result = priv->permissions))
804 gst_rtsp_permissions_ref (result);
805 g_mutex_unlock (&priv->lock);
811 * gst_rtsp_media_set_suspend_mode:
812 * @media: a #GstRTSPMedia
813 * @mode: the new #GstRTSPSuspendMode
815 * Control how @ media will be suspended after the SDP has been generated and
816 * after a PAUSE request has been performed.
818 * Media must be unprepared when setting the suspend mode.
821 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
823 GstRTSPMediaPrivate *priv;
825 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
829 g_rec_mutex_lock (&priv->state_lock);
830 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
832 priv->suspend_mode = mode;
833 g_rec_mutex_unlock (&priv->state_lock);
840 GST_WARNING ("media %p was prepared", media);
841 g_rec_mutex_unlock (&priv->state_lock);
846 * gst_rtsp_media_get_suspend_mode:
847 * @media: a #GstRTSPMedia
849 * Get how @media will be suspended.
851 * Returns: #GstRTSPSuspendMode.
854 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
856 GstRTSPMediaPrivate *priv;
857 GstRTSPSuspendMode res;
859 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
863 g_rec_mutex_lock (&priv->state_lock);
864 res = priv->suspend_mode;
865 g_rec_mutex_unlock (&priv->state_lock);
871 * gst_rtsp_media_set_shared:
872 * @media: a #GstRTSPMedia
873 * @shared: the new value
875 * Set or unset if the pipeline for @media can be shared will multiple clients.
876 * When @shared is %TRUE, client requests for this media will share the media
880 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
882 GstRTSPMediaPrivate *priv;
884 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
888 g_mutex_lock (&priv->lock);
889 priv->shared = shared;
890 g_mutex_unlock (&priv->lock);
894 * gst_rtsp_media_is_shared:
895 * @media: a #GstRTSPMedia
897 * Check if the pipeline for @media can be shared between multiple clients.
899 * Returns: %TRUE if the media can be shared between clients.
902 gst_rtsp_media_is_shared (GstRTSPMedia * media)
904 GstRTSPMediaPrivate *priv;
907 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
911 g_mutex_lock (&priv->lock);
913 g_mutex_unlock (&priv->lock);
919 * gst_rtsp_media_set_reusable:
920 * @media: a #GstRTSPMedia
921 * @reusable: the new value
923 * Set or unset if the pipeline for @media can be reused after the pipeline has
927 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
929 GstRTSPMediaPrivate *priv;
931 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
935 g_mutex_lock (&priv->lock);
936 priv->reusable = reusable;
937 g_mutex_unlock (&priv->lock);
941 * gst_rtsp_media_is_reusable:
942 * @media: a #GstRTSPMedia
944 * Check if the pipeline for @media can be reused after an unprepare.
946 * Returns: %TRUE if the media can be reused
949 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
951 GstRTSPMediaPrivate *priv;
954 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
958 g_mutex_lock (&priv->lock);
959 res = priv->reusable;
960 g_mutex_unlock (&priv->lock);
966 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
968 gst_rtsp_stream_set_profiles (stream, *profiles);
972 * gst_rtsp_media_set_profiles:
973 * @media: a #GstRTSPMedia
974 * @profiles: the new flags
976 * Configure the allowed lower transport for @media.
979 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
981 GstRTSPMediaPrivate *priv;
983 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
987 g_mutex_lock (&priv->lock);
988 priv->profiles = profiles;
989 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
990 g_mutex_unlock (&priv->lock);
994 * gst_rtsp_media_get_profiles:
995 * @media: a #GstRTSPMedia
997 * Get the allowed profiles of @media.
999 * Returns: a #GstRTSPProfile
1002 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1004 GstRTSPMediaPrivate *priv;
1007 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1011 g_mutex_lock (&priv->lock);
1012 res = priv->profiles;
1013 g_mutex_unlock (&priv->lock);
1019 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1021 gst_rtsp_stream_set_protocols (stream, *protocols);
1025 * gst_rtsp_media_set_protocols:
1026 * @media: a #GstRTSPMedia
1027 * @protocols: the new flags
1029 * Configure the allowed lower transport for @media.
1032 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1034 GstRTSPMediaPrivate *priv;
1036 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1040 g_mutex_lock (&priv->lock);
1041 priv->protocols = protocols;
1042 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1043 g_mutex_unlock (&priv->lock);
1047 * gst_rtsp_media_get_protocols:
1048 * @media: a #GstRTSPMedia
1050 * Get the allowed protocols of @media.
1052 * Returns: a #GstRTSPLowerTrans
1055 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1057 GstRTSPMediaPrivate *priv;
1058 GstRTSPLowerTrans res;
1060 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1061 GST_RTSP_LOWER_TRANS_UNKNOWN);
1065 g_mutex_lock (&priv->lock);
1066 res = priv->protocols;
1067 g_mutex_unlock (&priv->lock);
1073 * gst_rtsp_media_set_eos_shutdown:
1074 * @media: a #GstRTSPMedia
1075 * @eos_shutdown: the new value
1077 * Set or unset if an EOS event will be sent to the pipeline for @media before
1081 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1083 GstRTSPMediaPrivate *priv;
1085 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1089 g_mutex_lock (&priv->lock);
1090 priv->eos_shutdown = eos_shutdown;
1091 g_mutex_unlock (&priv->lock);
1095 * gst_rtsp_media_is_eos_shutdown:
1096 * @media: a #GstRTSPMedia
1098 * Check if the pipeline for @media will send an EOS down the pipeline before
1101 * Returns: %TRUE if the media will send EOS before unpreparing.
1104 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1106 GstRTSPMediaPrivate *priv;
1109 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1113 g_mutex_lock (&priv->lock);
1114 res = priv->eos_shutdown;
1115 g_mutex_unlock (&priv->lock);
1121 * gst_rtsp_media_set_buffer_size:
1122 * @media: a #GstRTSPMedia
1123 * @size: the new value
1125 * Set the kernel UDP buffer size.
1128 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1130 GstRTSPMediaPrivate *priv;
1133 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1135 GST_LOG_OBJECT (media, "set buffer size %u", size);
1139 g_mutex_lock (&priv->lock);
1140 priv->buffer_size = size;
1142 for (i = 0; i < priv->streams->len; i++) {
1143 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1144 gst_rtsp_stream_set_buffer_size (stream, size);
1146 g_mutex_unlock (&priv->lock);
1150 * gst_rtsp_media_get_buffer_size:
1151 * @media: a #GstRTSPMedia
1153 * Get the kernel UDP buffer size.
1155 * Returns: the kernel UDP buffer size.
1158 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1160 GstRTSPMediaPrivate *priv;
1163 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1167 g_mutex_lock (&priv->lock);
1168 res = priv->buffer_size;
1169 g_mutex_unlock (&priv->lock);
1175 * gst_rtsp_media_set_retransmission_time:
1176 * @media: a #GstRTSPMedia
1177 * @time: the new value
1179 * Set the amount of time to store retransmission packets.
1182 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1184 GstRTSPMediaPrivate *priv;
1187 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1189 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1193 g_mutex_lock (&priv->lock);
1194 priv->rtx_time = time;
1195 for (i = 0; i < priv->streams->len; i++) {
1196 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1198 gst_rtsp_stream_set_retransmission_time (stream, time);
1202 g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
1203 g_mutex_unlock (&priv->lock);
1207 * gst_rtsp_media_get_retransmission_time:
1208 * @media: a #GstRTSPMedia
1210 * Get the amount of time to store retransmission data.
1212 * Returns: the amount of time to store retransmission data.
1215 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1217 GstRTSPMediaPrivate *priv;
1220 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1224 g_mutex_unlock (&priv->lock);
1225 res = priv->rtx_time;
1226 g_mutex_unlock (&priv->lock);
1232 * gst_rtsp_media_set_latncy:
1233 * @media: a #GstRTSPMedia
1234 * @latency: latency in milliseconds
1236 * Configure the latency used for receiving media.
1239 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1241 GstRTSPMediaPrivate *priv;
1243 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1245 GST_LOG_OBJECT (media, "set latency %ums", latency);
1249 g_mutex_lock (&priv->lock);
1250 priv->latency = latency;
1252 g_object_set (priv->rtpbin, "latency", latency, NULL);
1253 g_mutex_unlock (&priv->lock);
1257 * gst_rtsp_media_get_latency:
1258 * @media: a #GstRTSPMedia
1260 * Get the latency that is used for receiving media.
1262 * Returns: latency in milliseconds
1265 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1267 GstRTSPMediaPrivate *priv;
1270 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1274 g_mutex_unlock (&priv->lock);
1275 res = priv->latency;
1276 g_mutex_unlock (&priv->lock);
1282 * gst_rtsp_media_use_time_provider:
1283 * @media: a #GstRTSPMedia
1284 * @time_provider: if a #GstNetTimeProvider should be used
1286 * Set @media to provide a #GstNetTimeProvider.
1289 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1291 GstRTSPMediaPrivate *priv;
1293 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1297 g_mutex_lock (&priv->lock);
1298 priv->time_provider = time_provider;
1299 g_mutex_unlock (&priv->lock);
1303 * gst_rtsp_media_is_time_provider:
1304 * @media: a #GstRTSPMedia
1306 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1308 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1310 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1313 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1315 GstRTSPMediaPrivate *priv;
1318 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1322 g_mutex_unlock (&priv->lock);
1323 res = priv->time_provider;
1324 g_mutex_unlock (&priv->lock);
1330 * gst_rtsp_media_set_address_pool:
1331 * @media: a #GstRTSPMedia
1332 * @pool: (transfer none): a #GstRTSPAddressPool
1334 * configure @pool to be used as the address pool of @media.
1337 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1338 GstRTSPAddressPool * pool)
1340 GstRTSPMediaPrivate *priv;
1341 GstRTSPAddressPool *old;
1343 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1347 GST_LOG_OBJECT (media, "set address pool %p", pool);
1349 g_mutex_lock (&priv->lock);
1350 if ((old = priv->pool) != pool)
1351 priv->pool = pool ? g_object_ref (pool) : NULL;
1354 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1356 g_mutex_unlock (&priv->lock);
1359 g_object_unref (old);
1363 * gst_rtsp_media_get_address_pool:
1364 * @media: a #GstRTSPMedia
1366 * Get the #GstRTSPAddressPool used as the address pool of @media.
1368 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1371 GstRTSPAddressPool *
1372 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1374 GstRTSPMediaPrivate *priv;
1375 GstRTSPAddressPool *result;
1377 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1381 g_mutex_lock (&priv->lock);
1382 if ((result = priv->pool))
1383 g_object_ref (result);
1384 g_mutex_unlock (&priv->lock);
1390 _find_payload_types (GstRTSPMedia * media)
1393 GQueue queue = G_QUEUE_INIT;
1395 n = media->priv->streams->len;
1396 for (i = 0; i < n; i++) {
1397 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1398 guint pt = gst_rtsp_stream_get_pt (stream);
1400 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1407 _next_available_pt (GList * payloads)
1411 for (i = 96; i <= 127; i++) {
1412 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1414 return GPOINTER_TO_UINT (i);
1421 * gst_rtsp_media_collect_streams:
1422 * @media: a #GstRTSPMedia
1424 * Find all payloader elements, they should be named pay\%d in the
1425 * element of @media, and create #GstRTSPStreams for them.
1427 * Collect all dynamic elements, named dynpay\%d, and add them to
1428 * the list of dynamic elements.
1430 * Find all depayloader elements, they should be named depay\%d in the
1431 * element of @media, and create #GstRTSPStreams for them.
1434 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1436 GstRTSPMediaPrivate *priv;
1437 GstElement *element, *elem;
1441 gboolean more_elem_remaining = TRUE;
1442 GstRTSPTransportMode mode = 0;
1444 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1447 element = priv->element;
1450 for (i = 0; more_elem_remaining; i++) {
1453 more_elem_remaining = FALSE;
1455 name = g_strdup_printf ("pay%d", i);
1456 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1458 GST_INFO ("found stream %d with payloader %p", i, elem);
1460 /* take the pad of the payloader */
1461 pad = gst_element_get_static_pad (elem, "src");
1463 /* find the real payload element in case elem is a GstBin */
1464 pay = find_payload_element (elem);
1466 /* create the stream */
1468 GST_WARNING ("could not find real payloader, using bin");
1469 gst_rtsp_media_create_stream (media, elem, pad);
1471 gst_rtsp_media_create_stream (media, pay, pad);
1472 gst_object_unref (pay);
1475 gst_object_unref (pad);
1476 gst_object_unref (elem);
1479 more_elem_remaining = TRUE;
1480 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1484 name = g_strdup_printf ("dynpay%d", i);
1485 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1486 /* a stream that will dynamically create pads to provide RTP packets */
1487 GST_INFO ("found dynamic element %d, %p", i, elem);
1489 g_mutex_lock (&priv->lock);
1490 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1491 g_mutex_unlock (&priv->lock);
1494 more_elem_remaining = TRUE;
1495 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1499 name = g_strdup_printf ("depay%d", i);
1500 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1501 GST_INFO ("found stream %d with depayloader %p", i, elem);
1503 /* take the pad of the payloader */
1504 pad = gst_element_get_static_pad (elem, "sink");
1505 /* create the stream */
1506 gst_rtsp_media_create_stream (media, elem, pad);
1507 gst_object_unref (pad);
1508 gst_object_unref (elem);
1511 more_elem_remaining = TRUE;
1512 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1518 if (priv->transport_mode != mode)
1519 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1520 priv->transport_mode, mode);
1525 * gst_rtsp_media_create_stream:
1526 * @media: a #GstRTSPMedia
1527 * @payloader: a #GstElement
1530 * Create a new stream in @media that provides RTP data on @pad.
1531 * @pad should be a pad of an element inside @media->element.
1533 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1537 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1540 GstRTSPMediaPrivate *priv;
1541 GstRTSPStream *stream;
1546 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1547 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1548 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1552 g_mutex_lock (&priv->lock);
1553 idx = priv->streams->len;
1555 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1557 if (GST_PAD_IS_SRC (pad))
1558 name = g_strdup_printf ("src_%u", idx);
1560 name = g_strdup_printf ("sink_%u", idx);
1562 ghostpad = gst_ghost_pad_new (name, pad);
1563 gst_pad_set_active (ghostpad, TRUE);
1564 gst_element_add_pad (priv->element, ghostpad);
1567 stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
1569 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1570 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1571 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1572 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
1573 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
1575 g_ptr_array_add (priv->streams, stream);
1577 if (GST_PAD_IS_SRC (pad)) {
1581 g_list_free (priv->payloads);
1582 priv->payloads = _find_payload_types (media);
1584 n = priv->streams->len;
1585 for (i = 0; i < n; i++) {
1586 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1587 guint rtx_pt = _next_available_pt (priv->payloads);
1590 GST_WARNING ("Ran out of space of dynamic payload types");
1594 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
1597 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
1600 g_mutex_unlock (&priv->lock);
1602 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1609 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1611 GstRTSPMediaPrivate *priv;
1616 g_mutex_lock (&priv->lock);
1617 /* remove the ghostpad */
1618 srcpad = gst_rtsp_stream_get_srcpad (stream);
1619 gst_element_remove_pad (priv->element, srcpad);
1620 gst_object_unref (srcpad);
1621 /* now remove the stream */
1622 g_object_ref (stream);
1623 g_ptr_array_remove (priv->streams, stream);
1624 g_mutex_unlock (&priv->lock);
1626 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1629 g_object_unref (stream);
1633 * gst_rtsp_media_n_streams:
1634 * @media: a #GstRTSPMedia
1636 * Get the number of streams in this media.
1638 * Returns: The number of streams.
1641 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1643 GstRTSPMediaPrivate *priv;
1646 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1650 g_mutex_lock (&priv->lock);
1651 res = priv->streams->len;
1652 g_mutex_unlock (&priv->lock);
1658 * gst_rtsp_media_get_stream:
1659 * @media: a #GstRTSPMedia
1660 * @idx: the stream index
1662 * Retrieve the stream with index @idx from @media.
1664 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
1665 * @idx or %NULL when a stream with that index did not exist.
1668 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1670 GstRTSPMediaPrivate *priv;
1673 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1677 g_mutex_lock (&priv->lock);
1678 if (idx < priv->streams->len)
1679 res = g_ptr_array_index (priv->streams, idx);
1682 g_mutex_unlock (&priv->lock);
1688 * gst_rtsp_media_find_stream:
1689 * @media: a #GstRTSPMedia
1690 * @control: the control of the stream
1692 * Find a stream in @media with @control as the control uri.
1694 * Returns: (nullable) (transfer none): the #GstRTSPStream with
1695 * control uri @control or %NULL when a stream with that control did
1699 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1701 GstRTSPMediaPrivate *priv;
1705 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1706 g_return_val_if_fail (control != NULL, NULL);
1712 g_mutex_lock (&priv->lock);
1713 for (i = 0; i < priv->streams->len; i++) {
1714 GstRTSPStream *test;
1716 test = g_ptr_array_index (priv->streams, i);
1717 if (gst_rtsp_stream_has_control (test, control)) {
1722 g_mutex_unlock (&priv->lock);
1727 /* called with state-lock */
1729 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1730 GstRTSPRangeUnit unit)
1732 return gst_rtsp_range_convert_units (range, unit);
1736 * gst_rtsp_media_get_range_string:
1737 * @media: a #GstRTSPMedia
1738 * @play: for the PLAY request
1739 * @unit: the unit to use for the string
1741 * Get the current range as a string. @media must be prepared with
1742 * gst_rtsp_media_prepare ().
1744 * Returns: (transfer full): The range as a string, g_free() after usage.
1747 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1748 GstRTSPRangeUnit unit)
1750 GstRTSPMediaClass *klass;
1751 GstRTSPMediaPrivate *priv;
1753 GstRTSPTimeRange range;
1755 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1756 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1757 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1761 g_rec_mutex_lock (&priv->state_lock);
1762 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1763 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1766 g_mutex_lock (&priv->lock);
1768 /* Update the range value with current position/duration */
1769 collect_media_stats (media);
1772 range = priv->range;
1774 if (!play && priv->n_active > 0) {
1775 range.min.type = GST_RTSP_TIME_NOW;
1776 range.min.seconds = -1;
1778 g_mutex_unlock (&priv->lock);
1779 g_rec_mutex_unlock (&priv->state_lock);
1781 if (!klass->convert_range (media, &range, unit))
1782 goto conversion_failed;
1784 result = gst_rtsp_range_to_string (&range);
1791 GST_WARNING ("media %p was not prepared", media);
1792 g_rec_mutex_unlock (&priv->state_lock);
1797 GST_WARNING ("range conversion to unit %d failed", unit);
1803 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1805 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1809 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1811 GstRTSPMediaPrivate *priv = media->priv;
1813 GST_DEBUG ("media %p set blocked %d", media, blocked);
1814 priv->blocked = blocked;
1815 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1819 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1821 GstRTSPMediaPrivate *priv = media->priv;
1823 g_mutex_lock (&priv->lock);
1824 priv->status = status;
1825 GST_DEBUG ("setting new status to %d", status);
1826 g_cond_broadcast (&priv->cond);
1827 g_mutex_unlock (&priv->lock);
1831 * gst_rtsp_media_get_status:
1832 * @media: a #GstRTSPMedia
1834 * Get the status of @media. When @media is busy preparing, this function waits
1835 * until @media is prepared or in error.
1837 * Returns: the status of @media.
1840 gst_rtsp_media_get_status (GstRTSPMedia * media)
1842 GstRTSPMediaPrivate *priv = media->priv;
1843 GstRTSPMediaStatus result;
1846 g_mutex_lock (&priv->lock);
1847 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1848 /* while we are preparing, wait */
1849 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1850 GST_DEBUG ("waiting for status change");
1851 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1852 GST_DEBUG ("timeout, assuming error status");
1853 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1856 /* could be success or error */
1857 result = priv->status;
1858 GST_DEBUG ("got status %d", result);
1859 g_mutex_unlock (&priv->lock);
1865 * gst_rtsp_media_seek:
1866 * @media: a #GstRTSPMedia
1867 * @range: (transfer none): a #GstRTSPTimeRange
1869 * Seek the pipeline of @media to @range. @media must be prepared with
1870 * gst_rtsp_media_prepare().
1872 * Returns: %TRUE on success.
1875 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1877 GstRTSPMediaClass *klass;
1878 GstRTSPMediaPrivate *priv;
1880 GstClockTime start, stop;
1881 GstSeekType start_type, stop_type;
1883 gint64 current_position;
1885 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1887 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1888 g_return_val_if_fail (range != NULL, FALSE);
1889 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1893 g_rec_mutex_lock (&priv->state_lock);
1894 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1897 /* Update the seekable state of the pipeline in case it changed */
1898 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
1899 /* TODO: Seeking for RECORD? */
1900 priv->seekable = FALSE;
1902 query = gst_query_new_seeking (GST_FORMAT_TIME);
1903 if (gst_element_query (priv->pipeline, query)) {
1908 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1909 priv->seekable = seekable;
1911 gst_query_unref (query);
1914 if (!priv->seekable)
1917 start_type = stop_type = GST_SEEK_TYPE_NONE;
1919 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1921 gst_rtsp_range_get_times (range, &start, &stop);
1923 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1924 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1925 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1926 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1928 current_position = -1;
1929 if (klass->query_position)
1930 klass->query_position (media, ¤t_position);
1931 GST_INFO ("current media position %" GST_TIME_FORMAT,
1932 GST_TIME_ARGS (current_position));
1934 if (start != GST_CLOCK_TIME_NONE)
1935 start_type = GST_SEEK_TYPE_SET;
1937 if (priv->range_stop == stop)
1938 stop = GST_CLOCK_TIME_NONE;
1939 else if (stop != GST_CLOCK_TIME_NONE)
1940 stop_type = GST_SEEK_TYPE_SET;
1942 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1945 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1946 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1948 /* depends on the current playing state of the pipeline. We might need to
1949 * queue this until we get EOS. */
1950 flags = GST_SEEK_FLAG_FLUSH;
1952 /* if range start was not supplied we must continue from current position.
1953 * but since we're doing a flushing seek, let us query the current position
1954 * so we end up at exactly the same position after the seek. */
1955 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
1956 if (current_position == -1) {
1957 GST_WARNING ("current position unknown");
1959 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
1960 GST_TIME_ARGS (current_position));
1961 start = current_position;
1962 start_type = GST_SEEK_TYPE_SET;
1963 flags |= GST_SEEK_FLAG_ACCURATE;
1966 /* only set keyframe flag when modifying start */
1967 if (start_type != GST_SEEK_TYPE_NONE)
1968 flags |= GST_SEEK_FLAG_KEY_UNIT;
1971 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
1972 GST_DEBUG ("not seeking because no position change");
1975 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
1977 media_streams_set_blocked (media, TRUE);
1979 /* FIXME, we only do forwards playback, no trick modes yet */
1980 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1981 flags, start_type, start, stop_type, stop);
1983 /* and block for the seek to complete */
1984 GST_INFO ("done seeking %d", res);
1988 g_rec_mutex_unlock (&priv->state_lock);
1990 /* wait until pipeline is prerolled again, this will also collect stats */
1991 if (!wait_preroll (media))
1992 goto preroll_failed;
1994 g_rec_mutex_lock (&priv->state_lock);
1995 GST_INFO ("prerolled again");
1998 GST_INFO ("no seek needed");
2001 g_rec_mutex_unlock (&priv->state_lock);
2008 g_rec_mutex_unlock (&priv->state_lock);
2009 GST_INFO ("media %p is not prepared", media);
2014 g_rec_mutex_unlock (&priv->state_lock);
2015 GST_INFO ("pipeline is not seekable");
2020 g_rec_mutex_unlock (&priv->state_lock);
2021 GST_WARNING ("conversion to npt not supported");
2026 g_rec_mutex_unlock (&priv->state_lock);
2027 GST_INFO ("seeking failed");
2028 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2033 GST_WARNING ("failed to preroll after seek");
2039 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2041 *blocked &= gst_rtsp_stream_is_blocking (stream);
2045 media_streams_blocking (GstRTSPMedia * media)
2047 gboolean blocking = TRUE;
2049 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2055 static GstStateChangeReturn
2056 set_state (GstRTSPMedia * media, GstState state)
2058 GstRTSPMediaPrivate *priv = media->priv;
2059 GstStateChangeReturn ret;
2061 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2063 ret = gst_element_set_state (priv->pipeline, state);
2068 GstStateChangeReturn
2069 gst_rtsp_media_set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2071 GstRTSPMediaPrivate *priv = media->priv;
2072 GstStateChangeReturn ret;
2074 GST_INFO ("set target state to %s for media %p",
2075 gst_element_state_get_name (state), media);
2076 priv->target_state = state;
2078 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2079 priv->target_state, NULL);
2082 ret = set_state (media, state);
2084 ret = GST_STATE_CHANGE_SUCCESS;
2089 /* called with state-lock */
2091 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2093 GstRTSPMediaPrivate *priv = media->priv;
2094 GstMessageType type;
2096 type = GST_MESSAGE_TYPE (message);
2099 case GST_MESSAGE_STATE_CHANGED:
2101 GstState old, new, pending;
2103 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2106 gst_message_parse_state_changed (message, &old, &new, &pending);
2108 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2109 gst_element_state_get_name (old), gst_element_state_get_name (new),
2110 gst_element_state_get_name (pending));
2111 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
2112 && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2113 GST_INFO ("%p: went to PAUSED, prepared now", media);
2114 collect_media_stats (media);
2116 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2117 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2122 case GST_MESSAGE_BUFFERING:
2126 gst_message_parse_buffering (message, &percent);
2128 /* no state management needed for live pipelines */
2132 if (percent == 100) {
2133 /* a 100% message means buffering is done */
2134 priv->buffering = FALSE;
2135 /* if the desired state is playing, go back */
2136 if (priv->target_state == GST_STATE_PLAYING) {
2137 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2138 set_state (media, GST_STATE_PLAYING);
2140 GST_INFO ("Buffering done");
2143 /* buffering busy */
2144 if (priv->buffering == FALSE) {
2145 if (priv->target_state == GST_STATE_PLAYING) {
2146 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2147 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2148 set_state (media, GST_STATE_PAUSED);
2150 GST_INFO ("Buffering ...");
2153 priv->buffering = TRUE;
2157 case GST_MESSAGE_LATENCY:
2159 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2162 case GST_MESSAGE_ERROR:
2167 gst_message_parse_error (message, &gerror, &debug);
2168 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2169 g_error_free (gerror);
2172 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2175 case GST_MESSAGE_WARNING:
2180 gst_message_parse_warning (message, &gerror, &debug);
2181 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2182 g_error_free (gerror);
2186 case GST_MESSAGE_ELEMENT:
2188 const GstStructure *s;
2190 s = gst_message_get_structure (message);
2191 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2192 GST_DEBUG ("media received blocking message");
2193 if (priv->blocked && media_streams_blocking (media)) {
2194 GST_DEBUG ("media is blocking");
2195 collect_media_stats (media);
2197 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2198 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2203 case GST_MESSAGE_STREAM_STATUS:
2205 case GST_MESSAGE_ASYNC_DONE:
2207 /* when we are dynamically adding pads, the addition of the udpsrc will
2208 * temporarily produce ASYNC_DONE messages. We have to ignore them and
2209 * wait for the final ASYNC_DONE after everything prerolled */
2210 GST_INFO ("%p: ignoring ASYNC_DONE", media);
2212 GST_INFO ("%p: got ASYNC_DONE", media);
2213 collect_media_stats (media);
2215 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2216 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2219 case GST_MESSAGE_EOS:
2220 GST_INFO ("%p: got EOS", media);
2222 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2223 GST_DEBUG ("shutting down after EOS");
2224 finish_unprepare (media);
2228 GST_INFO ("%p: got message type %d (%s)", media, type,
2229 gst_message_type_get_name (type));
2236 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2238 GstRTSPMediaPrivate *priv = media->priv;
2239 GstRTSPMediaClass *klass;
2242 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2244 g_rec_mutex_lock (&priv->state_lock);
2245 if (klass->handle_message)
2246 ret = klass->handle_message (media, message);
2249 g_rec_mutex_unlock (&priv->state_lock);
2255 watch_destroyed (GstRTSPMedia * media)
2257 GST_DEBUG_OBJECT (media, "source destroyed");
2258 g_object_unref (media);
2262 find_payload_element (GstElement * payloader)
2264 GstElement *pay = NULL;
2266 if (GST_IS_BIN (payloader)) {
2268 GValue item = { 0 };
2270 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2271 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2272 GstElement *element = (GstElement *) g_value_get_object (&item);
2273 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2277 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2281 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2282 pay = gst_object_ref (element);
2283 g_value_unset (&item);
2286 g_value_unset (&item);
2288 gst_iterator_free (iter);
2290 pay = g_object_ref (payloader);
2296 /* called from streaming threads */
2298 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2300 GstRTSPMediaPrivate *priv = media->priv;
2301 GstRTSPStream *stream;
2304 /* find the real payload element */
2305 pay = find_payload_element (element);
2306 stream = gst_rtsp_media_create_stream (media, pay, pad);
2307 gst_object_unref (pay);
2309 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2311 g_rec_mutex_lock (&priv->state_lock);
2312 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2315 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2317 /* we will be adding elements below that will cause ASYNC_DONE to be
2318 * posted in the bus. We want to ignore those messages until the
2319 * pipeline really prerolled. */
2320 priv->adding = TRUE;
2322 /* join the element in the PAUSED state because this callback is
2323 * called from the streaming thread and it is PAUSED */
2324 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2325 priv->rtpbin, GST_STATE_PAUSED)) {
2326 GST_WARNING ("failed to join bin element");
2329 priv->adding = FALSE;
2330 g_rec_mutex_unlock (&priv->state_lock);
2337 gst_rtsp_media_remove_stream (media, stream);
2338 g_rec_mutex_unlock (&priv->state_lock);
2339 GST_INFO ("ignore pad because we are not preparing");
2345 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2347 GstRTSPMediaPrivate *priv = media->priv;
2348 GstRTSPStream *stream;
2350 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2354 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2356 g_rec_mutex_lock (&priv->state_lock);
2357 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2358 g_rec_mutex_unlock (&priv->state_lock);
2360 gst_rtsp_media_remove_stream (media, stream);
2364 remove_fakesink (GstRTSPMediaPrivate * priv)
2366 GstElement *fakesink;
2368 g_mutex_lock (&priv->lock);
2369 if ((fakesink = priv->fakesink))
2370 gst_object_ref (fakesink);
2371 priv->fakesink = NULL;
2372 g_mutex_unlock (&priv->lock);
2375 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
2376 gst_element_set_state (fakesink, GST_STATE_NULL);
2377 gst_object_unref (fakesink);
2378 GST_INFO ("removed fakesink");
2383 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2385 GstRTSPMediaPrivate *priv = media->priv;
2387 GST_INFO ("no more pads");
2388 remove_fakesink (priv);
2391 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2393 struct _DynPaySignalHandlers
2395 gulong pad_added_handler;
2396 gulong pad_removed_handler;
2397 gulong no_more_pads_handler;
2401 default_start_preroll (GstRTSPMedia * media)
2403 GstRTSPMediaPrivate *priv = media->priv;
2404 GstStateChangeReturn ret;
2406 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2407 /* first go to PAUSED */
2408 ret = gst_rtsp_media_set_target_state (media, GST_STATE_PAUSED, TRUE);
2411 case GST_STATE_CHANGE_SUCCESS:
2412 GST_INFO ("SUCCESS state change for media %p", media);
2413 priv->seekable = TRUE;
2415 case GST_STATE_CHANGE_ASYNC:
2416 GST_INFO ("ASYNC state change for media %p", media);
2417 priv->seekable = TRUE;
2419 case GST_STATE_CHANGE_NO_PREROLL:
2420 /* we need to go to PLAYING */
2421 GST_INFO ("NO_PREROLL state change: live media %p", media);
2422 /* FIXME we disable seeking for live streams for now. We should perform a
2423 * seeking query in preroll instead */
2424 priv->seekable = FALSE;
2425 priv->is_live = TRUE;
2426 if (!(priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
2427 /* start blocked to make sure nothing goes to the sink */
2428 media_streams_set_blocked (media, TRUE);
2430 ret = set_state (media, GST_STATE_PLAYING);
2431 if (ret == GST_STATE_CHANGE_FAILURE)
2434 case GST_STATE_CHANGE_FAILURE:
2442 GST_WARNING ("failed to preroll pipeline");
2448 wait_preroll (GstRTSPMedia * media)
2450 GstRTSPMediaStatus status;
2452 GST_DEBUG ("wait to preroll pipeline");
2454 /* wait until pipeline is prerolled */
2455 status = gst_rtsp_media_get_status (media);
2456 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2457 goto preroll_failed;
2463 GST_WARNING ("failed to preroll pipeline");
2469 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
2471 GstRTSPMediaPrivate *priv = media->priv;
2472 GstRTSPStream *stream = NULL;
2475 g_mutex_lock (&priv->lock);
2476 for (i = 0; i < priv->streams->len; i++) {
2477 stream = g_ptr_array_index (priv->streams, i);
2479 if (sessid == gst_rtsp_stream_get_index (stream))
2482 g_mutex_unlock (&priv->lock);
2484 return gst_rtsp_stream_request_aux_sender (stream, sessid);
2488 default_start_prepare (GstRTSPMedia * media)
2490 GstRTSPMediaPrivate *priv = media->priv;
2491 GstRTSPMediaClass *klass;
2495 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2497 /* link streams we already have, other streams might appear when we have
2498 * dynamic elements */
2499 for (i = 0; i < priv->streams->len; i++) {
2500 GstRTSPStream *stream;
2502 stream = g_ptr_array_index (priv->streams, i);
2504 if (priv->rtx_time > 0) {
2505 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
2506 g_signal_connect (priv->rtpbin, "request-aux-sender",
2507 (GCallback) request_aux_sender, media);
2510 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2511 priv->rtpbin, GST_STATE_NULL)) {
2512 goto join_bin_failed;
2517 g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
2519 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2520 GstElement *elem = walk->data;
2521 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2523 GST_INFO ("adding callbacks for dynamic element %p", elem);
2525 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2526 (GCallback) pad_added_cb, media);
2527 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2528 (GCallback) pad_removed_cb, media);
2529 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2530 (GCallback) no_more_pads_cb, media);
2532 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2534 if (!priv->fakesink) {
2535 /* we add a fakesink here in order to make the state change async. We remove
2536 * the fakesink again in the no-more-pads callback. */
2537 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2538 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2542 if (klass->start_preroll)
2543 if(!klass->start_preroll (media))
2544 goto preroll_failed;
2550 GST_WARNING ("failed to join bin element");
2551 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2556 GST_WARNING ("failed to preroll pipeline");
2557 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2563 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2565 GstRTSPMediaPrivate *priv;
2566 GstRTSPMediaClass *klass;
2568 GMainContext *context;
2573 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2575 if (!klass->create_rtpbin)
2576 goto no_create_rtpbin;
2578 priv->rtpbin = klass->create_rtpbin (media);
2579 if (priv->rtpbin != NULL) {
2580 gboolean success = TRUE;
2582 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
2584 if (klass->setup_rtpbin)
2585 success = klass->setup_rtpbin (media, priv->rtpbin);
2587 if (success == FALSE) {
2588 gst_object_unref (priv->rtpbin);
2589 priv->rtpbin = NULL;
2592 if (priv->rtpbin == NULL)
2595 priv->thread = thread;
2596 context = (thread != NULL) ? (thread->context) : NULL;
2598 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2600 /* add the pipeline bus to our custom mainloop */
2601 priv->source = gst_bus_create_watch (bus);
2602 gst_object_unref (bus);
2604 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2605 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2607 priv->id = g_source_attach (priv->source, context);
2609 /* add stuff to the bin */
2610 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2612 /* do remainder in context */
2613 source = g_idle_source_new ();
2614 if(klass->start_prepare)
2615 g_source_set_callback (source, (GSourceFunc) klass->start_prepare,
2616 g_object_ref (media), (GDestroyNotify) g_object_unref);
2617 g_source_attach (source, context);
2618 g_source_unref (source);
2625 GST_ERROR ("no create_rtpbin function");
2626 g_critical ("no create_rtpbin vmethod function set");
2631 GST_WARNING ("no rtpbin element");
2632 g_warning ("failed to create element 'rtpbin', check your installation");
2638 * gst_rtsp_media_prepare:
2639 * @media: a #GstRTSPMedia
2640 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2641 * bus handler or %NULL
2643 * Prepare @media for streaming. This function will create the objects
2644 * to manage the streaming. A pipeline must have been set on @media with
2645 * gst_rtsp_media_take_pipeline().
2647 * It will preroll the pipeline and collect vital information about the streams
2648 * such as the duration.
2650 * Returns: %TRUE on success.
2653 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2655 GstRTSPMediaPrivate *priv;
2656 GstRTSPMediaClass *klass;
2658 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2662 g_rec_mutex_lock (&priv->state_lock);
2663 priv->prepare_count++;
2665 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2666 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2669 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2672 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2673 goto not_unprepared;
2675 if (!priv->reusable && priv->reused)
2678 GST_INFO ("preparing media %p", media);
2680 /* reset some variables */
2681 priv->is_live = FALSE;
2682 priv->seekable = FALSE;
2683 priv->buffering = FALSE;
2685 /* we're preparing now */
2686 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2688 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2689 if (klass->prepare) {
2690 if (!klass->prepare (media, thread))
2691 goto prepare_failed;
2695 g_rec_mutex_unlock (&priv->state_lock);
2697 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2698 * able to do this async so that we don't block the server thread. */
2699 if (!wait_preroll (media))
2700 goto preroll_failed;
2702 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2704 GST_INFO ("object %p is prerolled", media);
2711 /* we are not going to use the giving thread, so stop it. */
2713 gst_rtsp_thread_stop (thread);
2718 GST_LOG ("media %p was prepared", media);
2719 /* we are not going to use the giving thread, so stop it. */
2721 gst_rtsp_thread_stop (thread);
2722 g_rec_mutex_unlock (&priv->state_lock);
2728 /* we are not going to use the giving thread, so stop it. */
2730 gst_rtsp_thread_stop (thread);
2731 GST_WARNING ("media %p was not unprepared", media);
2732 priv->prepare_count--;
2733 g_rec_mutex_unlock (&priv->state_lock);
2738 /* we are not going to use the giving thread, so stop it. */
2740 gst_rtsp_thread_stop (thread);
2741 priv->prepare_count--;
2742 g_rec_mutex_unlock (&priv->state_lock);
2743 GST_WARNING ("can not reuse media %p", media);
2748 /* we are not going to use the giving thread, so stop it. */
2750 gst_rtsp_thread_stop (thread);
2751 priv->prepare_count--;
2752 g_rec_mutex_unlock (&priv->state_lock);
2753 GST_ERROR ("failed to prepare media");
2758 GST_WARNING ("failed to preroll pipeline");
2759 gst_rtsp_media_unprepare (media);
2764 /* must be called with state-lock */
2766 finish_unprepare (GstRTSPMedia * media)
2768 GstRTSPMediaPrivate *priv = media->priv;
2772 GST_DEBUG ("shutting down");
2774 /* release the lock on shutdown, otherwise pad_added_cb might try to
2775 * acquire the lock and then we deadlock */
2776 g_rec_mutex_unlock (&priv->state_lock);
2777 set_state (media, GST_STATE_NULL);
2778 g_rec_mutex_lock (&priv->state_lock);
2779 remove_fakesink (priv);
2781 for (i = 0; i < priv->streams->len; i++) {
2782 GstRTSPStream *stream;
2784 GST_INFO ("Removing elements of stream %d from pipeline", i);
2786 stream = g_ptr_array_index (priv->streams, i);
2788 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2791 /* remove the pad signal handlers */
2792 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2793 GstElement *elem = walk->data;
2794 DynPaySignalHandlers *handlers;
2797 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2798 g_assert (handlers != NULL);
2800 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2801 g_signal_handler_disconnect (G_OBJECT (elem),
2802 handlers->pad_removed_handler);
2803 g_signal_handler_disconnect (G_OBJECT (elem),
2804 handlers->no_more_pads_handler);
2806 g_slice_free (DynPaySignalHandlers, handlers);
2809 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2810 priv->rtpbin = NULL;
2813 gst_object_unref (priv->nettime);
2814 priv->nettime = NULL;
2816 priv->reused = TRUE;
2817 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
2819 /* when the media is not reusable, this will effectively unref the media and
2821 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2823 /* the source has the last ref to the media */
2825 GST_DEBUG ("destroy source");
2826 g_source_destroy (priv->source);
2827 g_source_unref (priv->source);
2830 GST_DEBUG ("stop thread");
2831 gst_rtsp_thread_stop (priv->thread);
2835 /* called with state-lock */
2837 default_unprepare (GstRTSPMedia * media)
2839 GstRTSPMediaPrivate *priv = media->priv;
2841 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
2843 if (priv->eos_shutdown) {
2844 GST_DEBUG ("sending EOS for shutdown");
2845 /* ref so that we don't disappear */
2846 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2847 /* we need to go to playing again for the EOS to propagate, normally in this
2848 * state, nothing is receiving data from us anymore so this is ok. */
2849 set_state (media, GST_STATE_PLAYING);
2851 finish_unprepare (media);
2857 * gst_rtsp_media_unprepare:
2858 * @media: a #GstRTSPMedia
2860 * Unprepare @media. After this call, the media should be prepared again before
2861 * it can be used again. If the media is set to be non-reusable, a new instance
2864 * Returns: %TRUE on success.
2867 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2869 GstRTSPMediaPrivate *priv;
2872 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2876 g_rec_mutex_lock (&priv->state_lock);
2877 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2878 goto was_unprepared;
2880 priv->prepare_count--;
2881 if (priv->prepare_count > 0)
2884 GST_INFO ("unprepare media %p", media);
2886 media_streams_set_blocked (media, FALSE);
2887 gst_rtsp_media_set_target_state (media, GST_STATE_NULL, FALSE);
2890 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2891 GstRTSPMediaClass *klass;
2893 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2894 if (klass->unprepare)
2895 success = klass->unprepare (media);
2897 finish_unprepare (media);
2899 g_rec_mutex_unlock (&priv->state_lock);
2905 g_rec_mutex_unlock (&priv->state_lock);
2906 GST_INFO ("media %p was already unprepared", media);
2911 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2912 g_rec_mutex_unlock (&priv->state_lock);
2917 /* should be called with state-lock */
2919 get_clock_unlocked (GstRTSPMedia * media)
2921 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2922 GST_DEBUG_OBJECT (media, "media was not prepared");
2925 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2929 * gst_rtsp_media_get_clock:
2930 * @media: a #GstRTSPMedia
2932 * Get the clock that is used by the pipeline in @media.
2934 * @media must be prepared before this method returns a valid clock object.
2936 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2939 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2942 GstRTSPMediaPrivate *priv;
2944 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2948 g_rec_mutex_lock (&priv->state_lock);
2949 clock = get_clock_unlocked (media);
2950 g_rec_mutex_unlock (&priv->state_lock);
2956 * gst_rtsp_media_get_base_time:
2957 * @media: a #GstRTSPMedia
2959 * Get the base_time that is used by the pipeline in @media.
2961 * @media must be prepared before this method returns a valid base_time.
2963 * Returns: the base_time used by @media.
2966 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2968 GstClockTime result;
2969 GstRTSPMediaPrivate *priv;
2971 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2975 g_rec_mutex_lock (&priv->state_lock);
2976 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2979 result = gst_element_get_base_time (media->priv->pipeline);
2980 g_rec_mutex_unlock (&priv->state_lock);
2987 g_rec_mutex_unlock (&priv->state_lock);
2988 GST_DEBUG_OBJECT (media, "media was not prepared");
2989 return GST_CLOCK_TIME_NONE;
2994 * gst_rtsp_media_get_time_provider:
2995 * @media: a #GstRTSPMedia
2996 * @address: (allow-none): an address or %NULL
2997 * @port: a port or 0
2999 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3000 * will listen on @address and @port for client time requests.
3002 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3004 GstNetTimeProvider *
3005 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3008 GstRTSPMediaPrivate *priv;
3009 GstNetTimeProvider *provider = NULL;
3011 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3015 g_rec_mutex_lock (&priv->state_lock);
3016 if (priv->time_provider) {
3017 if ((provider = priv->nettime) == NULL) {
3020 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3021 provider = gst_net_time_provider_new (clock, address, port);
3022 gst_object_unref (clock);
3024 priv->nettime = provider;
3028 g_rec_mutex_unlock (&priv->state_lock);
3031 gst_object_ref (provider);
3037 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3039 return gst_rtsp_sdp_from_media (sdp, info, media);
3043 * gst_rtsp_media_setup_sdp:
3044 * @media: a #GstRTSPMedia
3045 * @sdp: (transfer none): a #GstSDPMessage
3046 * @info: (transfer none): a #GstSDPInfo
3048 * Add @media specific info to @sdp. @info is used to configure the connection
3049 * information in the SDP.
3051 * Returns: TRUE on success.
3054 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3057 GstRTSPMediaPrivate *priv;
3058 GstRTSPMediaClass *klass;
3061 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3062 g_return_val_if_fail (sdp != NULL, FALSE);
3063 g_return_val_if_fail (info != NULL, FALSE);
3067 g_rec_mutex_lock (&priv->state_lock);
3069 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3071 if (!klass->setup_sdp)
3074 res = klass->setup_sdp (media, sdp, info);
3076 g_rec_mutex_unlock (&priv->state_lock);
3083 g_rec_mutex_unlock (&priv->state_lock);
3084 GST_ERROR ("no setup_sdp function");
3085 g_critical ("no setup_sdp vmethod function set");
3090 static const gchar *
3091 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
3100 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
3103 if (sscanf (attr, "%d ", &val) != 1)
3112 #define PARSE_INT(p, del, res) \
3115 p = strstr (p, del); \
3125 #define PARSE_STRING(p, del, res) \
3128 p = strstr (p, del); \
3140 #define SKIP_SPACES(p) \
3141 while (*p && g_ascii_isspace (*p)) \
3146 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
3149 parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
3150 gint * rate, gchar ** params)
3154 p = (gchar *) rtpmap;
3156 PARSE_INT (p, " ", *payload);
3164 PARSE_STRING (p, "/", *name);
3165 if (*name == NULL) {
3166 GST_DEBUG ("no rate, name %s", p);
3167 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
3168 * streams seem to omit the rate. */
3175 p = strstr (p, "/");
3193 * Mapping of caps to and from SDP fields:
3195 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
3196 * a=framesize:<payload> <width>-<height>
3197 * a=fmtp:<payload> <param>[=<value>];...
3200 media_to_caps (gint pt, const GstSDPMedia * media)
3203 const gchar *rtpmap;
3205 const gchar *framesize;
3208 gchar *params = NULL;
3214 /* get and parse rtpmap */
3215 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
3218 ret = parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
3220 g_warning ("error parsing rtpmap, ignoring");
3224 /* dynamic payloads need rtpmap or we fail */
3225 if (rtpmap == NULL && pt >= 96)
3228 /* check if we have a rate, if not, we need to look up the rate from the
3229 * default rates based on the payload types. */
3231 const GstRTPPayloadInfo *info;
3233 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
3234 /* dynamic types, use media and encoding_name */
3235 tmp = g_ascii_strdown (media->media, -1);
3236 info = gst_rtp_payload_info_for_name (tmp, name);
3239 /* static types, use payload type */
3240 info = gst_rtp_payload_info_for_pt (pt);
3244 if ((rate = info->clock_rate) == 0)
3247 /* we fail if we cannot find one */
3252 tmp = g_ascii_strdown (media->media, -1);
3253 caps = gst_caps_new_simple ("application/x-unknown",
3254 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
3256 s = gst_caps_get_structure (caps, 0);
3258 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
3260 /* encoding name must be upper case */
3262 tmp = g_ascii_strup (name, -1);
3263 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
3267 /* params must be lower case */
3268 if (params != NULL) {
3269 tmp = g_ascii_strdown (params, -1);
3270 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
3274 /* parse optional fmtp: field */
3275 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
3281 /* p is now of the format <payload> <param>[=<value>];... */
3282 PARSE_INT (p, " ", payload);
3283 if (payload != -1 && payload == pt) {
3287 /* <param>[=<value>] are separated with ';' */
3288 pairs = g_strsplit (p, ";", 0);
3289 for (i = 0; pairs[i]; i++) {
3291 const gchar *val, *key;
3293 const gchar *reserved_keys[] =
3294 { "media", "payload", "clock-rate", "encoding-name",
3298 /* the key may not have a '=', the value can have other '='s */
3299 valpos = strstr (pairs[i], "=");
3301 /* we have a '=' and thus a value, remove the '=' with \0 */
3303 /* value is everything between '=' and ';'. We split the pairs at ;
3304 * boundaries so we can take the remainder of the value. Some servers
3305 * put spaces around the value which we strip off here. Alternatively
3306 * we could strip those spaces in the depayloaders should these spaces
3307 * actually carry any meaning in the future. */
3308 val = g_strstrip (valpos + 1);
3310 /* simple <param>;.. is translated into <param>=1;... */
3313 /* strip the key of spaces, convert key to lowercase but not the value. */
3314 key = g_strstrip (pairs[i]);
3316 /* skip keys from the fmtp, which we already use ourselves for the
3317 * caps. Some software is adding random things like clock-rate into
3318 * the fmtp, and we would otherwise here set a string-typed clock-rate
3319 * in the caps... and thus fail to create valid RTP caps
3321 for (j = 0; j < G_N_ELEMENTS (reserved_keys); j++) {
3322 if (g_ascii_strcasecmp (reserved_keys[j], key) == 0) {
3328 if (strlen (key) > 1) {
3329 tmp = g_ascii_strdown (key, -1);
3330 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
3338 /* parse framesize: field */
3339 if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
3342 /* p is now of the format <payload> <width>-<height> */
3343 p = (gchar *) framesize;
3345 PARSE_INT (p, " ", payload);
3346 if (payload != -1 && payload == pt) {
3347 gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
3355 g_warning ("rtpmap type not given for dynamic payload %d", pt);
3360 g_warning ("rate unknown for payload type %d", pt);
3366 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
3368 gboolean res = FALSE;
3371 GstMIKEYMessage *msg;
3372 const GstMIKEYPayload *payload;
3373 const gchar *srtp_cipher;
3374 const gchar *srtp_auth;
3380 p = orig_value = g_strdup (keymgmt);
3384 g_free (orig_value);
3388 PARSE_STRING (p, " ", kmpid);
3389 if (kmpid == NULL || !g_str_equal (kmpid, "mikey")) {
3390 g_free (orig_value);
3393 data = g_base64_decode (p, &size);
3395 g_free (orig_value); /* Don't need this any more */
3401 msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
3406 srtp_cipher = "aes-128-icm";
3407 srtp_auth = "hmac-sha1-80";
3409 /* check the Security policy if any */
3410 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
3411 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
3414 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
3417 len = gst_mikey_payload_sp_get_n_params (payload);
3418 for (i = 0; i < len; i++) {
3419 const GstMIKEYPayloadSPParam *param =
3420 gst_mikey_payload_sp_get_param (payload, i);
3422 switch (param->type) {
3423 case GST_MIKEY_SP_SRTP_ENC_ALG:
3424 switch (param->val[0]) {
3426 srtp_cipher = "null";
3430 srtp_cipher = "aes-128-icm";
3436 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
3437 switch (param->val[0]) {
3438 case AES_128_KEY_LEN:
3439 srtp_cipher = "aes-128-icm";
3441 case AES_256_KEY_LEN:
3442 srtp_cipher = "aes-256-icm";
3448 case GST_MIKEY_SP_SRTP_AUTH_ALG:
3449 switch (param->val[0]) {
3455 srtp_auth = "hmac-sha1-80";
3461 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
3462 switch (param->val[0]) {
3463 case HMAC_32_KEY_LEN:
3464 srtp_auth = "hmac-sha1-32";
3466 case HMAC_80_KEY_LEN:
3467 srtp_auth = "hmac-sha1-80";
3473 case GST_MIKEY_SP_SRTP_SRTP_ENC:
3475 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
3483 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
3486 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
3487 const GstMIKEYPayload *sub;
3488 GstMIKEYPayloadKeyData *pkd;
3491 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
3494 if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
3497 if (sub->type != GST_MIKEY_PT_KEY_DATA)
3500 pkd = (GstMIKEYPayloadKeyData *) sub;
3502 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
3504 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
3507 gst_caps_set_simple (caps,
3508 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
3509 "srtp-auth", G_TYPE_STRING, srtp_auth,
3510 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
3511 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
3515 gst_mikey_message_unref (msg);
3521 * Mapping SDP attributes to caps
3523 * prepend 'a-' to IANA registered sdp attributes names
3524 * (ie: not prefixed with 'x-') in order to avoid
3525 * collision with gstreamer standard caps properties names
3528 sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
3530 if (attributes->len > 0) {
3534 s = gst_caps_get_structure (caps, 0);
3536 for (i = 0; i < attributes->len; i++) {
3537 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
3538 gchar *tofree, *key;
3542 /* skip some of the attribute we already handle */
3543 if (!strcmp (key, "fmtp"))
3545 if (!strcmp (key, "rtpmap"))
3547 if (!strcmp (key, "control"))
3549 if (!strcmp (key, "range"))
3551 if (!strcmp (key, "framesize"))
3553 if (g_str_equal (key, "key-mgmt")) {
3554 parse_keymgmt (attr->value, caps);
3558 /* string must be valid UTF8 */
3559 if (!g_utf8_validate (attr->value, -1, NULL))
3562 if (!g_str_has_prefix (key, "x-"))
3563 tofree = key = g_strdup_printf ("a-%s", key);
3567 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
3568 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
3575 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3577 GstRTSPMediaPrivate *priv = media->priv;
3580 medias_len = gst_sdp_message_medias_len (sdp);
3581 if (medias_len != priv->streams->len) {
3582 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3583 priv->streams->len, medias_len);
3587 for (i = 0; i < medias_len; i++) {
3588 const gchar *proto, *media_type;
3589 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3590 GstRTSPStream *stream;
3591 gint j, formats_len;
3592 const gchar *control;
3593 GstRTSPProfile profile, profiles;
3595 stream = g_ptr_array_index (priv->streams, i);
3597 /* TODO: Should we do something with the other SDP information? */
3600 proto = gst_sdp_media_get_proto (sdp_media);
3601 if (proto == NULL) {
3602 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3606 if (g_str_equal (proto, "RTP/AVP")) {
3607 media_type = "application/x-rtp";
3608 profile = GST_RTSP_PROFILE_AVP;
3609 } else if (g_str_equal (proto, "RTP/SAVP")) {
3610 media_type = "application/x-srtp";
3611 profile = GST_RTSP_PROFILE_SAVP;
3612 } else if (g_str_equal (proto, "RTP/AVPF")) {
3613 media_type = "application/x-rtp";
3614 profile = GST_RTSP_PROFILE_AVPF;
3615 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3616 media_type = "application/x-srtp";
3617 profile = GST_RTSP_PROFILE_SAVPF;
3619 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3623 profiles = gst_rtsp_stream_get_profiles (stream);
3624 if ((profiles & profile) == 0) {
3625 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3629 formats_len = gst_sdp_media_formats_len (sdp_media);
3630 for (j = 0; j < formats_len; j++) {
3635 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3637 GST_DEBUG (" looking at %d pt: %d", j, pt);
3640 caps = media_to_caps (pt, sdp_media);
3642 GST_WARNING (" skipping pt %d without caps", pt);
3646 /* do some tweaks */
3647 GST_DEBUG ("mapping sdp session level attributes to caps");
3648 sdp_attributes_to_caps (sdp->attributes, caps);
3649 GST_DEBUG ("mapping sdp media level attributes to caps");
3650 sdp_attributes_to_caps (sdp_media->attributes, caps);
3652 s = gst_caps_get_structure (caps, 0);
3653 gst_structure_set_name (s, media_type);
3655 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3656 gst_caps_unref (caps);
3659 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3661 gst_rtsp_stream_set_control (stream, control);
3669 * gst_rtsp_media_handle_sdp:
3670 * @media: a #GstRTSPMedia
3671 * @sdp: (transfer none): a #GstSDPMessage
3673 * Configure an SDP on @media for receiving streams
3675 * Returns: TRUE on success.
3678 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3680 GstRTSPMediaPrivate *priv;
3681 GstRTSPMediaClass *klass;
3684 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3685 g_return_val_if_fail (sdp != NULL, FALSE);
3689 g_rec_mutex_lock (&priv->state_lock);
3691 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3693 if (!klass->handle_sdp)
3696 res = klass->handle_sdp (media, sdp);
3698 g_rec_mutex_unlock (&priv->state_lock);
3705 g_rec_mutex_unlock (&priv->state_lock);
3706 GST_ERROR ("no handle_sdp function");
3707 g_critical ("no handle_sdp vmethod function set");
3713 do_set_seqnum (GstRTSPStream * stream)
3716 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3717 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3720 /* call with state_lock */
3722 default_suspend (GstRTSPMedia * media)
3724 GstRTSPMediaPrivate *priv = media->priv;
3725 GstStateChangeReturn ret;
3726 gboolean unblock = FALSE;
3728 switch (priv->suspend_mode) {
3729 case GST_RTSP_SUSPEND_MODE_NONE:
3730 GST_DEBUG ("media %p no suspend", media);
3732 case GST_RTSP_SUSPEND_MODE_PAUSE:
3733 GST_DEBUG ("media %p suspend to PAUSED", media);
3734 ret = gst_rtsp_media_set_target_state (media, GST_STATE_PAUSED, TRUE);
3735 if (ret == GST_STATE_CHANGE_FAILURE)
3739 case GST_RTSP_SUSPEND_MODE_RESET:
3740 GST_DEBUG ("media %p suspend to NULL", media);
3741 ret = gst_rtsp_media_set_target_state (media, GST_STATE_NULL, TRUE);
3742 if (ret == GST_STATE_CHANGE_FAILURE)
3744 /* Because payloader needs to set the sequence number as
3745 * monotonic, we need to preserve the sequence number
3746 * after pause. (otherwise going from pause to play, which
3747 * is actually from NULL to PLAY will create a new sequence
3749 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3756 /* let the streams do the state changes freely, if any */
3758 media_streams_set_blocked (media, FALSE);
3765 GST_WARNING ("failed changing pipeline's state for media %p", media);
3771 * gst_rtsp_media_suspend:
3772 * @media: a #GstRTSPMedia
3774 * Suspend @media. The state of the pipeline managed by @media is set to
3775 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3776 * with gst_rtsp_media_unsuspend()
3778 * @media must be prepared with gst_rtsp_media_prepare();
3780 * Returns: %TRUE on success.
3783 gst_rtsp_media_suspend (GstRTSPMedia * media)
3785 GstRTSPMediaPrivate *priv = media->priv;
3786 GstRTSPMediaClass *klass;
3788 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3790 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3792 g_rec_mutex_lock (&priv->state_lock);
3793 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3796 /* don't attempt to suspend when something is busy */
3797 if (priv->n_active > 0)
3800 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3801 if (klass->suspend) {
3802 if (!klass->suspend (media))
3803 goto suspend_failed;
3806 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3808 g_rec_mutex_unlock (&priv->state_lock);
3815 g_rec_mutex_unlock (&priv->state_lock);
3816 GST_WARNING ("media %p was not prepared", media);
3821 g_rec_mutex_unlock (&priv->state_lock);
3822 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3823 GST_WARNING ("failed to suspend media %p", media);
3828 /* call with state_lock */
3830 default_unsuspend (GstRTSPMedia * media)
3832 GstRTSPMediaPrivate *priv = media->priv;
3833 GstRTSPMediaClass *klass;
3835 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3837 switch (priv->suspend_mode) {
3838 case GST_RTSP_SUSPEND_MODE_NONE:
3839 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3841 case GST_RTSP_SUSPEND_MODE_PAUSE:
3842 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3844 case GST_RTSP_SUSPEND_MODE_RESET:
3846 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3847 if (klass->start_preroll)
3848 if(!klass->start_preroll (media))
3850 g_rec_mutex_unlock (&priv->state_lock);
3852 if (!wait_preroll (media))
3853 goto preroll_failed;
3855 g_rec_mutex_lock (&priv->state_lock);
3866 GST_WARNING ("failed to preroll pipeline");
3871 GST_WARNING ("failed to preroll pipeline");
3877 * gst_rtsp_media_unsuspend:
3878 * @media: a #GstRTSPMedia
3880 * Unsuspend @media if it was in a suspended state. This method does nothing
3881 * when the media was not in the suspended state.
3883 * Returns: %TRUE on success.
3886 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
3888 GstRTSPMediaPrivate *priv = media->priv;
3889 GstRTSPMediaClass *klass;
3891 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3893 g_rec_mutex_lock (&priv->state_lock);
3894 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3897 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3898 if (klass->unsuspend) {
3899 if (!klass->unsuspend (media))
3900 goto unsuspend_failed;
3904 g_rec_mutex_unlock (&priv->state_lock);
3911 g_rec_mutex_unlock (&priv->state_lock);
3912 GST_WARNING ("failed to unsuspend media %p", media);
3913 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3918 /* must be called with state-lock */
3920 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
3922 GstRTSPMediaPrivate *priv = media->priv;
3924 if (state == GST_STATE_NULL) {
3925 gst_rtsp_media_unprepare (media);
3927 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
3928 gst_rtsp_media_set_target_state (media, state, FALSE);
3929 /* when we are buffering, don't update the state yet, this will be done
3930 * when buffering finishes */
3931 if (priv->buffering) {
3932 GST_INFO ("Buffering busy, delay state change");
3934 if (state == GST_STATE_PLAYING)
3935 /* make sure pads are not blocking anymore when going to PLAYING */
3936 media_streams_set_blocked (media, FALSE);
3938 set_state (media, state);
3940 /* and suspend after pause */
3941 if (state == GST_STATE_PAUSED)
3942 gst_rtsp_media_suspend (media);
3948 * gst_rtsp_media_set_pipeline_state:
3949 * @media: a #GstRTSPMedia
3950 * @state: the target state of the pipeline
3952 * Set the state of the pipeline managed by @media to @state
3955 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
3957 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
3959 g_rec_mutex_lock (&media->priv->state_lock);
3960 media_set_pipeline_state_locked (media, state);
3961 g_rec_mutex_unlock (&media->priv->state_lock);
3965 * gst_rtsp_media_set_state:
3966 * @media: a #GstRTSPMedia
3967 * @state: the target state of the media
3968 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
3969 * a #GPtrArray of #GstRTSPStreamTransport pointers
3971 * Set the state of @media to @state and for the transports in @transports.
3973 * @media must be prepared with gst_rtsp_media_prepare();
3975 * Returns: %TRUE on success.
3978 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
3979 GPtrArray * transports)
3981 GstRTSPMediaPrivate *priv;
3983 gboolean activate, deactivate, do_state;
3986 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3987 g_return_val_if_fail (transports != NULL, FALSE);
3991 g_rec_mutex_lock (&priv->state_lock);
3992 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
3994 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
3995 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3998 /* NULL and READY are the same */
3999 if (state == GST_STATE_READY)
4000 state = GST_STATE_NULL;
4002 activate = deactivate = FALSE;
4004 GST_INFO ("going to state %s media %p, target state %s",
4005 gst_element_state_get_name (state), media,
4006 gst_element_state_get_name (priv->target_state));
4009 case GST_STATE_NULL:
4010 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4011 if (priv->target_state >= GST_STATE_PAUSED)
4014 case GST_STATE_PAUSED:
4015 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4016 if (priv->target_state == GST_STATE_PLAYING)
4019 case GST_STATE_PLAYING:
4020 /* we're going to PLAYING, activate */
4026 old_active = priv->n_active;
4028 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4029 activate, deactivate);
4030 for (i = 0; i < transports->len; i++) {
4031 GstRTSPStreamTransport *trans;
4033 /* we need a non-NULL entry in the array */
4034 trans = g_ptr_array_index (transports, i);
4039 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4041 } else if (deactivate) {
4042 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4047 /* we just activated the first media, do the playing state change */
4048 if (old_active == 0 && activate)
4050 /* if we have no more active media, do the downward state changes */
4051 else if (priv->n_active == 0)
4056 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4059 if (priv->target_state != state) {
4061 media_set_pipeline_state_locked (media, state);
4063 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4067 /* remember where we are */
4068 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4069 old_active != priv->n_active))
4070 collect_media_stats (media);
4072 g_rec_mutex_unlock (&priv->state_lock);
4079 GST_WARNING ("media %p was not prepared", media);
4080 g_rec_mutex_unlock (&priv->state_lock);
4085 GST_WARNING ("media %p in error status while changing to state %d",
4087 if (state == GST_STATE_NULL) {
4088 for (i = 0; i < transports->len; i++) {
4089 GstRTSPStreamTransport *trans;
4091 /* we need a non-NULL entry in the array */
4092 trans = g_ptr_array_index (transports, i);
4096 gst_rtsp_stream_transport_set_active (trans, FALSE);
4100 g_rec_mutex_unlock (&priv->state_lock);
4106 * gst_rtsp_media_set_transport_mode:
4107 * @media: a #GstRTSPMedia
4108 * @mode: the new value
4110 * Sets if the media pipeline can work in PLAY or RECORD mode
4113 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4114 GstRTSPTransportMode mode)
4116 GstRTSPMediaPrivate *priv;
4118 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4122 g_mutex_lock (&priv->lock);
4123 priv->transport_mode = mode;
4124 g_mutex_unlock (&priv->lock);
4128 * gst_rtsp_media_get_transport_mode:
4129 * @media: a #GstRTSPMedia
4131 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4133 * Returns: The transport mode.
4135 GstRTSPTransportMode
4136 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4138 GstRTSPMediaPrivate *priv;
4139 GstRTSPTransportMode res;
4141 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4145 g_mutex_lock (&priv->lock);
4146 res = priv->transport_mode;
4147 g_mutex_unlock (&priv->lock);