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;
105 gboolean stop_on_disconnect;
108 GRecMutex state_lock; /* locking order: state lock, lock */
109 GPtrArray *streams; /* protected by lock */
110 GList *dynamic; /* protected by lock */
111 GstRTSPMediaStatus status; /* protected by lock */
116 /* the pipeline for the media */
117 GstElement *pipeline;
118 GstElement *fakesink; /* protected by lock */
121 GstRTSPThread *thread;
123 gboolean time_provider;
124 GstNetTimeProvider *nettime;
129 GstState target_state;
131 /* RTP session manager */
134 /* the range of media */
135 GstRTSPTimeRange range; /* protected by lock */
136 GstClockTime range_start;
137 GstClockTime range_stop;
139 GList *payloads; /* protected by lock */
140 GstClockTime rtx_time; /* protected by lock */
141 guint latency; /* protected by lock */
144 #define DEFAULT_SHARED FALSE
145 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
146 #define DEFAULT_REUSABLE FALSE
147 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
148 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
149 GST_RTSP_LOWER_TRANS_TCP
150 #define DEFAULT_EOS_SHUTDOWN FALSE
151 #define DEFAULT_BUFFER_SIZE 0x80000
152 #define DEFAULT_TIME_PROVIDER FALSE
153 #define DEFAULT_LATENCY 200
154 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
155 #define DEFAULT_STOP_ON_DISCONNECT TRUE
157 /* define to dump received RTCP packets */
174 PROP_STOP_ON_DISCONNECT,
181 SIGNAL_REMOVED_STREAM,
189 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
190 #define GST_CAT_DEFAULT rtsp_media_debug
192 static void gst_rtsp_media_get_property (GObject * object, guint propid,
193 GValue * value, GParamSpec * pspec);
194 static void gst_rtsp_media_set_property (GObject * object, guint propid,
195 const GValue * value, GParamSpec * pspec);
196 static void gst_rtsp_media_finalize (GObject * obj);
198 static gboolean default_handle_message (GstRTSPMedia * media,
199 GstMessage * message);
200 static void finish_unprepare (GstRTSPMedia * media);
201 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
202 static gboolean default_unprepare (GstRTSPMedia * media);
203 static gboolean default_suspend (GstRTSPMedia * media);
204 static gboolean default_unsuspend (GstRTSPMedia * media);
205 static gboolean default_convert_range (GstRTSPMedia * media,
206 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
207 static gboolean default_query_position (GstRTSPMedia * media,
209 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
210 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
211 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
213 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
215 static gboolean wait_preroll (GstRTSPMedia * media);
217 static GstElement *find_payload_element (GstElement * payloader);
219 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
221 #define C_ENUM(v) ((gint) v)
224 gst_rtsp_suspend_mode_get_type (void)
227 static const GEnumValue values[] = {
228 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
229 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
231 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
236 if (g_once_init_enter (&id)) {
237 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
238 g_once_init_leave (&id, tmp);
243 #define C_FLAGS(v) ((guint) v)
246 gst_rtsp_transport_mode_get_type (void)
249 static const GFlagsValue values[] = {
250 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
252 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
257 if (g_once_init_enter (&id)) {
258 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
259 g_once_init_leave (&id, tmp);
264 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
267 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
269 GObjectClass *gobject_class;
271 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
273 gobject_class = G_OBJECT_CLASS (klass);
275 gobject_class->get_property = gst_rtsp_media_get_property;
276 gobject_class->set_property = gst_rtsp_media_set_property;
277 gobject_class->finalize = gst_rtsp_media_finalize;
279 g_object_class_install_property (gobject_class, PROP_SHARED,
280 g_param_spec_boolean ("shared", "Shared",
281 "If this media pipeline can be shared", DEFAULT_SHARED,
282 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
284 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
285 g_param_spec_enum ("suspend-mode", "Suspend Mode",
286 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
287 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
289 g_object_class_install_property (gobject_class, PROP_REUSABLE,
290 g_param_spec_boolean ("reusable", "Reusable",
291 "If this media pipeline can be reused after an unprepare",
292 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
294 g_object_class_install_property (gobject_class, PROP_PROFILES,
295 g_param_spec_flags ("profiles", "Profiles",
296 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
297 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
299 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
300 g_param_spec_flags ("protocols", "Protocols",
301 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
302 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
304 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
305 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
306 "Send an EOS event to the pipeline before unpreparing",
307 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
309 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
310 g_param_spec_uint ("buffer-size", "Buffer Size",
311 "The kernel UDP buffer size to use", 0, G_MAXUINT,
312 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
314 g_object_class_install_property (gobject_class, PROP_ELEMENT,
315 g_param_spec_object ("element", "The Element",
316 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
317 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
319 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
320 g_param_spec_boolean ("time-provider", "Time Provider",
321 "Use a NetTimeProvider for clients",
322 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
324 g_object_class_install_property (gobject_class, PROP_LATENCY,
325 g_param_spec_uint ("latency", "Latency",
326 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
327 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
329 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
330 g_param_spec_flags ("transport-mode", "Transport Mode",
331 "If this media pipeline can be used for PLAY or RECORD",
332 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
333 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
335 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
336 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
337 "If this media pipeline should be stopped "
338 "when a client disconnects without TEARDOWN",
339 DEFAULT_STOP_ON_DISCONNECT,
340 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
342 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
343 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
344 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
345 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
347 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
348 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
349 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
350 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
351 GST_TYPE_RTSP_STREAM);
353 gst_rtsp_media_signals[SIGNAL_PREPARED] =
354 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
355 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
356 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
358 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
359 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
360 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
361 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
363 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
364 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
365 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
366 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
368 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
369 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
370 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
371 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
373 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
375 klass->handle_message = default_handle_message;
376 klass->prepare = default_prepare;
377 klass->unprepare = default_unprepare;
378 klass->suspend = default_suspend;
379 klass->unsuspend = default_unsuspend;
380 klass->convert_range = default_convert_range;
381 klass->query_position = default_query_position;
382 klass->query_stop = default_query_stop;
383 klass->create_rtpbin = default_create_rtpbin;
384 klass->setup_sdp = default_setup_sdp;
385 klass->handle_sdp = default_handle_sdp;
389 gst_rtsp_media_init (GstRTSPMedia * media)
391 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
395 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
396 g_mutex_init (&priv->lock);
397 g_cond_init (&priv->cond);
398 g_rec_mutex_init (&priv->state_lock);
400 priv->shared = DEFAULT_SHARED;
401 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
402 priv->reusable = DEFAULT_REUSABLE;
403 priv->profiles = DEFAULT_PROFILES;
404 priv->protocols = DEFAULT_PROTOCOLS;
405 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
406 priv->buffer_size = DEFAULT_BUFFER_SIZE;
407 priv->time_provider = DEFAULT_TIME_PROVIDER;
408 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
409 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
413 gst_rtsp_media_finalize (GObject * obj)
415 GstRTSPMediaPrivate *priv;
418 media = GST_RTSP_MEDIA (obj);
421 GST_INFO ("finalize media %p", media);
423 if (priv->permissions)
424 gst_rtsp_permissions_unref (priv->permissions);
426 g_ptr_array_unref (priv->streams);
428 g_list_free_full (priv->dynamic, gst_object_unref);
431 gst_object_unref (priv->pipeline);
433 gst_object_unref (priv->nettime);
434 gst_object_unref (priv->element);
436 g_object_unref (priv->pool);
438 g_list_free (priv->payloads);
439 g_mutex_clear (&priv->lock);
440 g_cond_clear (&priv->cond);
441 g_rec_mutex_clear (&priv->state_lock);
443 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
447 gst_rtsp_media_get_property (GObject * object, guint propid,
448 GValue * value, GParamSpec * pspec)
450 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
454 g_value_set_object (value, media->priv->element);
457 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
459 case PROP_SUSPEND_MODE:
460 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
463 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
466 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
469 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
471 case PROP_EOS_SHUTDOWN:
472 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
474 case PROP_BUFFER_SIZE:
475 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
477 case PROP_TIME_PROVIDER:
478 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
481 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
483 case PROP_TRANSPORT_MODE:
484 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
486 case PROP_STOP_ON_DISCONNECT:
487 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
490 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
495 gst_rtsp_media_set_property (GObject * object, guint propid,
496 const GValue * value, GParamSpec * pspec)
498 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
502 media->priv->element = g_value_get_object (value);
503 gst_object_ref_sink (media->priv->element);
506 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
508 case PROP_SUSPEND_MODE:
509 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
512 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
515 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
518 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
520 case PROP_EOS_SHUTDOWN:
521 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
523 case PROP_BUFFER_SIZE:
524 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
526 case PROP_TIME_PROVIDER:
527 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
530 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
532 case PROP_TRANSPORT_MODE:
533 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
535 case PROP_STOP_ON_DISCONNECT:
536 gst_rtsp_media_set_stop_on_disconnect (media,
537 g_value_get_boolean (value));
540 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
548 } DoQueryPositionData;
551 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
555 if (gst_rtsp_stream_query_position (stream, &tmp)) {
556 data->position = MAX (data->position, tmp);
562 default_query_position (GstRTSPMedia * media, gint64 * position)
564 GstRTSPMediaPrivate *priv;
565 DoQueryPositionData data;
572 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
574 *position = data.position;
586 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
590 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
591 data->stop = MAX (data->stop, tmp);
597 default_query_stop (GstRTSPMedia * media, gint64 * stop)
599 GstRTSPMediaPrivate *priv;
600 DoQueryStopData data;
607 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
615 default_create_rtpbin (GstRTSPMedia * media)
619 rtpbin = gst_element_factory_make ("rtpbin", NULL);
624 /* must be called with state lock */
626 collect_media_stats (GstRTSPMedia * media)
628 GstRTSPMediaPrivate *priv = media->priv;
629 gint64 position = 0, stop = -1;
631 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
632 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
635 priv->range.unit = GST_RTSP_RANGE_NPT;
637 GST_INFO ("collect media stats");
640 priv->range.min.type = GST_RTSP_TIME_NOW;
641 priv->range.min.seconds = -1;
642 priv->range_start = -1;
643 priv->range.max.type = GST_RTSP_TIME_END;
644 priv->range.max.seconds = -1;
645 priv->range_stop = -1;
647 GstRTSPMediaClass *klass;
650 klass = GST_RTSP_MEDIA_GET_CLASS (media);
652 /* get the position */
654 if (klass->query_position)
655 ret = klass->query_position (media, &position);
658 GST_INFO ("position query failed");
662 /* get the current segment stop */
664 if (klass->query_stop)
665 ret = klass->query_stop (media, &stop);
668 GST_INFO ("stop query failed");
672 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
673 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
675 if (position == -1) {
676 priv->range.min.type = GST_RTSP_TIME_NOW;
677 priv->range.min.seconds = -1;
678 priv->range_start = -1;
680 priv->range.min.type = GST_RTSP_TIME_SECONDS;
681 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
682 priv->range_start = position;
685 priv->range.max.type = GST_RTSP_TIME_END;
686 priv->range.max.seconds = -1;
687 priv->range_stop = -1;
689 priv->range.max.type = GST_RTSP_TIME_SECONDS;
690 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
691 priv->range_stop = stop;
697 * gst_rtsp_media_new:
698 * @element: (transfer full): a #GstElement
700 * Create a new #GstRTSPMedia instance. @element is the bin element that
701 * provides the different streams. The #GstRTSPMedia object contains the
702 * element to produce RTP data for one or more related (audio/video/..)
705 * Ownership is taken of @element.
707 * Returns: (transfer full): a new #GstRTSPMedia object.
710 gst_rtsp_media_new (GstElement * element)
712 GstRTSPMedia *result;
714 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
716 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
722 * gst_rtsp_media_get_element:
723 * @media: a #GstRTSPMedia
725 * Get the element that was used when constructing @media.
727 * Returns: (transfer full): a #GstElement. Unref after usage.
730 gst_rtsp_media_get_element (GstRTSPMedia * media)
732 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
734 return gst_object_ref (media->priv->element);
738 * gst_rtsp_media_take_pipeline:
739 * @media: a #GstRTSPMedia
740 * @pipeline: (transfer full): a #GstPipeline
742 * Set @pipeline as the #GstPipeline for @media. Ownership is
743 * taken of @pipeline.
746 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
748 GstRTSPMediaPrivate *priv;
750 GstNetTimeProvider *nettime;
752 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
753 g_return_if_fail (GST_IS_PIPELINE (pipeline));
757 g_mutex_lock (&priv->lock);
758 old = priv->pipeline;
759 priv->pipeline = GST_ELEMENT_CAST (pipeline);
760 nettime = priv->nettime;
761 priv->nettime = NULL;
762 g_mutex_unlock (&priv->lock);
765 gst_object_unref (old);
768 gst_object_unref (nettime);
770 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
774 * gst_rtsp_media_set_permissions:
775 * @media: a #GstRTSPMedia
776 * @permissions: (transfer none): a #GstRTSPPermissions
778 * Set @permissions on @media.
781 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
782 GstRTSPPermissions * permissions)
784 GstRTSPMediaPrivate *priv;
786 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
790 g_mutex_lock (&priv->lock);
791 if (priv->permissions)
792 gst_rtsp_permissions_unref (priv->permissions);
793 if ((priv->permissions = permissions))
794 gst_rtsp_permissions_ref (permissions);
795 g_mutex_unlock (&priv->lock);
799 * gst_rtsp_media_get_permissions:
800 * @media: a #GstRTSPMedia
802 * Get the permissions object from @media.
804 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
807 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
809 GstRTSPMediaPrivate *priv;
810 GstRTSPPermissions *result;
812 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
816 g_mutex_lock (&priv->lock);
817 if ((result = priv->permissions))
818 gst_rtsp_permissions_ref (result);
819 g_mutex_unlock (&priv->lock);
825 * gst_rtsp_media_set_suspend_mode:
826 * @media: a #GstRTSPMedia
827 * @mode: the new #GstRTSPSuspendMode
829 * Control how @ media will be suspended after the SDP has been generated and
830 * after a PAUSE request has been performed.
832 * Media must be unprepared when setting the suspend mode.
835 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
837 GstRTSPMediaPrivate *priv;
839 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
843 g_rec_mutex_lock (&priv->state_lock);
844 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
846 priv->suspend_mode = mode;
847 g_rec_mutex_unlock (&priv->state_lock);
854 GST_WARNING ("media %p was prepared", media);
855 g_rec_mutex_unlock (&priv->state_lock);
860 * gst_rtsp_media_get_suspend_mode:
861 * @media: a #GstRTSPMedia
863 * Get how @media will be suspended.
865 * Returns: #GstRTSPSuspendMode.
868 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
870 GstRTSPMediaPrivate *priv;
871 GstRTSPSuspendMode res;
873 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
877 g_rec_mutex_lock (&priv->state_lock);
878 res = priv->suspend_mode;
879 g_rec_mutex_unlock (&priv->state_lock);
885 * gst_rtsp_media_set_shared:
886 * @media: a #GstRTSPMedia
887 * @shared: the new value
889 * Set or unset if the pipeline for @media can be shared will multiple clients.
890 * When @shared is %TRUE, client requests for this media will share the media
894 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
896 GstRTSPMediaPrivate *priv;
898 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
902 g_mutex_lock (&priv->lock);
903 priv->shared = shared;
904 g_mutex_unlock (&priv->lock);
908 * gst_rtsp_media_is_shared:
909 * @media: a #GstRTSPMedia
911 * Check if the pipeline for @media can be shared between multiple clients.
913 * Returns: %TRUE if the media can be shared between clients.
916 gst_rtsp_media_is_shared (GstRTSPMedia * media)
918 GstRTSPMediaPrivate *priv;
921 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
925 g_mutex_lock (&priv->lock);
927 g_mutex_unlock (&priv->lock);
933 * gst_rtsp_media_set_reusable:
934 * @media: a #GstRTSPMedia
935 * @reusable: the new value
937 * Set or unset if the pipeline for @media can be reused after the pipeline has
941 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
943 GstRTSPMediaPrivate *priv;
945 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
949 g_mutex_lock (&priv->lock);
950 priv->reusable = reusable;
951 g_mutex_unlock (&priv->lock);
955 * gst_rtsp_media_is_reusable:
956 * @media: a #GstRTSPMedia
958 * Check if the pipeline for @media can be reused after an unprepare.
960 * Returns: %TRUE if the media can be reused
963 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
965 GstRTSPMediaPrivate *priv;
968 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
972 g_mutex_lock (&priv->lock);
973 res = priv->reusable;
974 g_mutex_unlock (&priv->lock);
980 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
982 gst_rtsp_stream_set_profiles (stream, *profiles);
986 * gst_rtsp_media_set_profiles:
987 * @media: a #GstRTSPMedia
988 * @profiles: the new flags
990 * Configure the allowed lower transport for @media.
993 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
995 GstRTSPMediaPrivate *priv;
997 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1001 g_mutex_lock (&priv->lock);
1002 priv->profiles = profiles;
1003 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1004 g_mutex_unlock (&priv->lock);
1008 * gst_rtsp_media_get_profiles:
1009 * @media: a #GstRTSPMedia
1011 * Get the allowed profiles of @media.
1013 * Returns: a #GstRTSPProfile
1016 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1018 GstRTSPMediaPrivate *priv;
1021 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1025 g_mutex_lock (&priv->lock);
1026 res = priv->profiles;
1027 g_mutex_unlock (&priv->lock);
1033 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1035 gst_rtsp_stream_set_protocols (stream, *protocols);
1039 * gst_rtsp_media_set_protocols:
1040 * @media: a #GstRTSPMedia
1041 * @protocols: the new flags
1043 * Configure the allowed lower transport for @media.
1046 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1048 GstRTSPMediaPrivate *priv;
1050 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1054 g_mutex_lock (&priv->lock);
1055 priv->protocols = protocols;
1056 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1057 g_mutex_unlock (&priv->lock);
1061 * gst_rtsp_media_get_protocols:
1062 * @media: a #GstRTSPMedia
1064 * Get the allowed protocols of @media.
1066 * Returns: a #GstRTSPLowerTrans
1069 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1071 GstRTSPMediaPrivate *priv;
1072 GstRTSPLowerTrans res;
1074 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1075 GST_RTSP_LOWER_TRANS_UNKNOWN);
1079 g_mutex_lock (&priv->lock);
1080 res = priv->protocols;
1081 g_mutex_unlock (&priv->lock);
1087 * gst_rtsp_media_set_eos_shutdown:
1088 * @media: a #GstRTSPMedia
1089 * @eos_shutdown: the new value
1091 * Set or unset if an EOS event will be sent to the pipeline for @media before
1095 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1097 GstRTSPMediaPrivate *priv;
1099 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1103 g_mutex_lock (&priv->lock);
1104 priv->eos_shutdown = eos_shutdown;
1105 g_mutex_unlock (&priv->lock);
1109 * gst_rtsp_media_is_eos_shutdown:
1110 * @media: a #GstRTSPMedia
1112 * Check if the pipeline for @media will send an EOS down the pipeline before
1115 * Returns: %TRUE if the media will send EOS before unpreparing.
1118 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1120 GstRTSPMediaPrivate *priv;
1123 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1127 g_mutex_lock (&priv->lock);
1128 res = priv->eos_shutdown;
1129 g_mutex_unlock (&priv->lock);
1135 * gst_rtsp_media_set_buffer_size:
1136 * @media: a #GstRTSPMedia
1137 * @size: the new value
1139 * Set the kernel UDP buffer size.
1142 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1144 GstRTSPMediaPrivate *priv;
1147 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1149 GST_LOG_OBJECT (media, "set buffer size %u", size);
1153 g_mutex_lock (&priv->lock);
1154 priv->buffer_size = size;
1156 for (i = 0; i < priv->streams->len; i++) {
1157 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1158 gst_rtsp_stream_set_buffer_size (stream, size);
1160 g_mutex_unlock (&priv->lock);
1164 * gst_rtsp_media_get_buffer_size:
1165 * @media: a #GstRTSPMedia
1167 * Get the kernel UDP buffer size.
1169 * Returns: the kernel UDP buffer size.
1172 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1174 GstRTSPMediaPrivate *priv;
1177 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1181 g_mutex_lock (&priv->lock);
1182 res = priv->buffer_size;
1183 g_mutex_unlock (&priv->lock);
1189 * gst_rtsp_media_set_stop_on_disconnect:
1190 * @media: a #GstRTSPMedia
1191 * @stop_on_disconnect: the new value
1193 * Set or unset if the pipeline for @media should be stopped when a
1194 * client disconnects without sending TEARDOWN.
1197 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1198 gboolean stop_on_disconnect)
1200 GstRTSPMediaPrivate *priv;
1202 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1206 g_mutex_lock (&priv->lock);
1207 priv->stop_on_disconnect = stop_on_disconnect;
1208 g_mutex_unlock (&priv->lock);
1212 * gst_rtsp_media_is_stop_on_disconnect:
1213 * @media: a #GstRTSPMedia
1215 * Check if the pipeline for @media will be stopped when a client disconnects
1216 * without sending TEARDOWN.
1218 * Returns: %TRUE if the media will be stopped when a client disconnects
1219 * without sending TEARDOWN.
1222 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1224 GstRTSPMediaPrivate *priv;
1227 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1231 g_mutex_lock (&priv->lock);
1232 res = priv->stop_on_disconnect;
1233 g_mutex_unlock (&priv->lock);
1239 * gst_rtsp_media_set_retransmission_time:
1240 * @media: a #GstRTSPMedia
1241 * @time: the new value
1243 * Set the amount of time to store retransmission packets.
1246 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1248 GstRTSPMediaPrivate *priv;
1251 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1253 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1257 g_mutex_lock (&priv->lock);
1258 priv->rtx_time = time;
1259 for (i = 0; i < priv->streams->len; i++) {
1260 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1262 gst_rtsp_stream_set_retransmission_time (stream, time);
1266 g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
1267 g_mutex_unlock (&priv->lock);
1271 * gst_rtsp_media_get_retransmission_time:
1272 * @media: a #GstRTSPMedia
1274 * Get the amount of time to store retransmission data.
1276 * Returns: the amount of time to store retransmission data.
1279 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1281 GstRTSPMediaPrivate *priv;
1284 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1288 g_mutex_unlock (&priv->lock);
1289 res = priv->rtx_time;
1290 g_mutex_unlock (&priv->lock);
1296 * gst_rtsp_media_set_latncy:
1297 * @media: a #GstRTSPMedia
1298 * @latency: latency in milliseconds
1300 * Configure the latency used for receiving media.
1303 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1305 GstRTSPMediaPrivate *priv;
1307 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1309 GST_LOG_OBJECT (media, "set latency %ums", latency);
1313 g_mutex_lock (&priv->lock);
1314 priv->latency = latency;
1316 g_object_set (priv->rtpbin, "latency", latency, NULL);
1317 g_mutex_unlock (&priv->lock);
1321 * gst_rtsp_media_get_latency:
1322 * @media: a #GstRTSPMedia
1324 * Get the latency that is used for receiving media.
1326 * Returns: latency in milliseconds
1329 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1331 GstRTSPMediaPrivate *priv;
1334 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1338 g_mutex_unlock (&priv->lock);
1339 res = priv->latency;
1340 g_mutex_unlock (&priv->lock);
1346 * gst_rtsp_media_use_time_provider:
1347 * @media: a #GstRTSPMedia
1348 * @time_provider: if a #GstNetTimeProvider should be used
1350 * Set @media to provide a #GstNetTimeProvider.
1353 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1355 GstRTSPMediaPrivate *priv;
1357 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1361 g_mutex_lock (&priv->lock);
1362 priv->time_provider = time_provider;
1363 g_mutex_unlock (&priv->lock);
1367 * gst_rtsp_media_is_time_provider:
1368 * @media: a #GstRTSPMedia
1370 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1372 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1374 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1377 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1379 GstRTSPMediaPrivate *priv;
1382 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1386 g_mutex_unlock (&priv->lock);
1387 res = priv->time_provider;
1388 g_mutex_unlock (&priv->lock);
1394 * gst_rtsp_media_set_address_pool:
1395 * @media: a #GstRTSPMedia
1396 * @pool: (transfer none): a #GstRTSPAddressPool
1398 * configure @pool to be used as the address pool of @media.
1401 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1402 GstRTSPAddressPool * pool)
1404 GstRTSPMediaPrivate *priv;
1405 GstRTSPAddressPool *old;
1407 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1411 GST_LOG_OBJECT (media, "set address pool %p", pool);
1413 g_mutex_lock (&priv->lock);
1414 if ((old = priv->pool) != pool)
1415 priv->pool = pool ? g_object_ref (pool) : NULL;
1418 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1420 g_mutex_unlock (&priv->lock);
1423 g_object_unref (old);
1427 * gst_rtsp_media_get_address_pool:
1428 * @media: a #GstRTSPMedia
1430 * Get the #GstRTSPAddressPool used as the address pool of @media.
1432 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1435 GstRTSPAddressPool *
1436 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1438 GstRTSPMediaPrivate *priv;
1439 GstRTSPAddressPool *result;
1441 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1445 g_mutex_lock (&priv->lock);
1446 if ((result = priv->pool))
1447 g_object_ref (result);
1448 g_mutex_unlock (&priv->lock);
1454 _find_payload_types (GstRTSPMedia * media)
1457 GQueue queue = G_QUEUE_INIT;
1459 n = media->priv->streams->len;
1460 for (i = 0; i < n; i++) {
1461 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1462 guint pt = gst_rtsp_stream_get_pt (stream);
1464 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1471 _next_available_pt (GList * payloads)
1475 for (i = 96; i <= 127; i++) {
1476 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1478 return GPOINTER_TO_UINT (i);
1485 * gst_rtsp_media_collect_streams:
1486 * @media: a #GstRTSPMedia
1488 * Find all payloader elements, they should be named pay\%d in the
1489 * element of @media, and create #GstRTSPStreams for them.
1491 * Collect all dynamic elements, named dynpay\%d, and add them to
1492 * the list of dynamic elements.
1494 * Find all depayloader elements, they should be named depay\%d in the
1495 * element of @media, and create #GstRTSPStreams for them.
1498 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1500 GstRTSPMediaPrivate *priv;
1501 GstElement *element, *elem;
1505 gboolean more_elem_remaining = TRUE;
1506 GstRTSPTransportMode mode = 0;
1508 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1511 element = priv->element;
1514 for (i = 0; more_elem_remaining; i++) {
1517 more_elem_remaining = FALSE;
1519 name = g_strdup_printf ("pay%d", i);
1520 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1522 GST_INFO ("found stream %d with payloader %p", i, elem);
1524 /* take the pad of the payloader */
1525 pad = gst_element_get_static_pad (elem, "src");
1527 /* find the real payload element in case elem is a GstBin */
1528 pay = find_payload_element (elem);
1530 /* create the stream */
1532 GST_WARNING ("could not find real payloader, using bin");
1533 gst_rtsp_media_create_stream (media, elem, pad);
1535 gst_rtsp_media_create_stream (media, pay, pad);
1536 gst_object_unref (pay);
1539 gst_object_unref (pad);
1540 gst_object_unref (elem);
1543 more_elem_remaining = TRUE;
1544 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1548 name = g_strdup_printf ("dynpay%d", i);
1549 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1550 /* a stream that will dynamically create pads to provide RTP packets */
1551 GST_INFO ("found dynamic element %d, %p", i, elem);
1553 g_mutex_lock (&priv->lock);
1554 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1555 g_mutex_unlock (&priv->lock);
1558 more_elem_remaining = TRUE;
1559 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1563 name = g_strdup_printf ("depay%d", i);
1564 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1565 GST_INFO ("found stream %d with depayloader %p", i, elem);
1567 /* take the pad of the payloader */
1568 pad = gst_element_get_static_pad (elem, "sink");
1569 /* create the stream */
1570 gst_rtsp_media_create_stream (media, elem, pad);
1571 gst_object_unref (pad);
1572 gst_object_unref (elem);
1575 more_elem_remaining = TRUE;
1576 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1582 if (priv->transport_mode != mode)
1583 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1584 priv->transport_mode, mode);
1589 * gst_rtsp_media_create_stream:
1590 * @media: a #GstRTSPMedia
1591 * @payloader: a #GstElement
1594 * Create a new stream in @media that provides RTP data on @pad.
1595 * @pad should be a pad of an element inside @media->element.
1597 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1601 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1604 GstRTSPMediaPrivate *priv;
1605 GstRTSPStream *stream;
1610 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1611 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1612 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1616 g_mutex_lock (&priv->lock);
1617 idx = priv->streams->len;
1619 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1621 if (GST_PAD_IS_SRC (pad))
1622 name = g_strdup_printf ("src_%u", idx);
1624 name = g_strdup_printf ("sink_%u", idx);
1626 ghostpad = gst_ghost_pad_new (name, pad);
1627 gst_pad_set_active (ghostpad, TRUE);
1628 gst_element_add_pad (priv->element, ghostpad);
1631 stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
1633 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1634 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1635 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1636 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
1637 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
1639 g_ptr_array_add (priv->streams, stream);
1641 if (GST_PAD_IS_SRC (pad)) {
1645 g_list_free (priv->payloads);
1646 priv->payloads = _find_payload_types (media);
1648 n = priv->streams->len;
1649 for (i = 0; i < n; i++) {
1650 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1651 guint rtx_pt = _next_available_pt (priv->payloads);
1654 GST_WARNING ("Ran out of space of dynamic payload types");
1658 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
1661 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
1664 g_mutex_unlock (&priv->lock);
1666 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1673 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1675 GstRTSPMediaPrivate *priv;
1680 g_mutex_lock (&priv->lock);
1681 /* remove the ghostpad */
1682 srcpad = gst_rtsp_stream_get_srcpad (stream);
1683 gst_element_remove_pad (priv->element, srcpad);
1684 gst_object_unref (srcpad);
1685 /* now remove the stream */
1686 g_object_ref (stream);
1687 g_ptr_array_remove (priv->streams, stream);
1688 g_mutex_unlock (&priv->lock);
1690 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1693 g_object_unref (stream);
1697 * gst_rtsp_media_n_streams:
1698 * @media: a #GstRTSPMedia
1700 * Get the number of streams in this media.
1702 * Returns: The number of streams.
1705 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1707 GstRTSPMediaPrivate *priv;
1710 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1714 g_mutex_lock (&priv->lock);
1715 res = priv->streams->len;
1716 g_mutex_unlock (&priv->lock);
1722 * gst_rtsp_media_get_stream:
1723 * @media: a #GstRTSPMedia
1724 * @idx: the stream index
1726 * Retrieve the stream with index @idx from @media.
1728 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
1729 * @idx or %NULL when a stream with that index did not exist.
1732 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1734 GstRTSPMediaPrivate *priv;
1737 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1741 g_mutex_lock (&priv->lock);
1742 if (idx < priv->streams->len)
1743 res = g_ptr_array_index (priv->streams, idx);
1746 g_mutex_unlock (&priv->lock);
1752 * gst_rtsp_media_find_stream:
1753 * @media: a #GstRTSPMedia
1754 * @control: the control of the stream
1756 * Find a stream in @media with @control as the control uri.
1758 * Returns: (nullable) (transfer none): the #GstRTSPStream with
1759 * control uri @control or %NULL when a stream with that control did
1763 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1765 GstRTSPMediaPrivate *priv;
1769 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1770 g_return_val_if_fail (control != NULL, NULL);
1776 g_mutex_lock (&priv->lock);
1777 for (i = 0; i < priv->streams->len; i++) {
1778 GstRTSPStream *test;
1780 test = g_ptr_array_index (priv->streams, i);
1781 if (gst_rtsp_stream_has_control (test, control)) {
1786 g_mutex_unlock (&priv->lock);
1791 /* called with state-lock */
1793 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1794 GstRTSPRangeUnit unit)
1796 return gst_rtsp_range_convert_units (range, unit);
1800 * gst_rtsp_media_get_range_string:
1801 * @media: a #GstRTSPMedia
1802 * @play: for the PLAY request
1803 * @unit: the unit to use for the string
1805 * Get the current range as a string. @media must be prepared with
1806 * gst_rtsp_media_prepare ().
1808 * Returns: (transfer full): The range as a string, g_free() after usage.
1811 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1812 GstRTSPRangeUnit unit)
1814 GstRTSPMediaClass *klass;
1815 GstRTSPMediaPrivate *priv;
1817 GstRTSPTimeRange range;
1819 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1820 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1821 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1825 g_rec_mutex_lock (&priv->state_lock);
1826 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1827 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1830 g_mutex_lock (&priv->lock);
1832 /* Update the range value with current position/duration */
1833 collect_media_stats (media);
1836 range = priv->range;
1838 if (!play && priv->n_active > 0) {
1839 range.min.type = GST_RTSP_TIME_NOW;
1840 range.min.seconds = -1;
1842 g_mutex_unlock (&priv->lock);
1843 g_rec_mutex_unlock (&priv->state_lock);
1845 if (!klass->convert_range (media, &range, unit))
1846 goto conversion_failed;
1848 result = gst_rtsp_range_to_string (&range);
1855 GST_WARNING ("media %p was not prepared", media);
1856 g_rec_mutex_unlock (&priv->state_lock);
1861 GST_WARNING ("range conversion to unit %d failed", unit);
1867 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1869 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1873 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1875 GstRTSPMediaPrivate *priv = media->priv;
1877 GST_DEBUG ("media %p set blocked %d", media, blocked);
1878 priv->blocked = blocked;
1879 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1883 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1885 GstRTSPMediaPrivate *priv = media->priv;
1887 g_mutex_lock (&priv->lock);
1888 priv->status = status;
1889 GST_DEBUG ("setting new status to %d", status);
1890 g_cond_broadcast (&priv->cond);
1891 g_mutex_unlock (&priv->lock);
1895 * gst_rtsp_media_get_status:
1896 * @media: a #GstRTSPMedia
1898 * Get the status of @media. When @media is busy preparing, this function waits
1899 * until @media is prepared or in error.
1901 * Returns: the status of @media.
1904 gst_rtsp_media_get_status (GstRTSPMedia * media)
1906 GstRTSPMediaPrivate *priv = media->priv;
1907 GstRTSPMediaStatus result;
1910 g_mutex_lock (&priv->lock);
1911 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1912 /* while we are preparing, wait */
1913 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1914 GST_DEBUG ("waiting for status change");
1915 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1916 GST_DEBUG ("timeout, assuming error status");
1917 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1920 /* could be success or error */
1921 result = priv->status;
1922 GST_DEBUG ("got status %d", result);
1923 g_mutex_unlock (&priv->lock);
1929 * gst_rtsp_media_seek:
1930 * @media: a #GstRTSPMedia
1931 * @range: (transfer none): a #GstRTSPTimeRange
1933 * Seek the pipeline of @media to @range. @media must be prepared with
1934 * gst_rtsp_media_prepare().
1936 * Returns: %TRUE on success.
1939 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1941 GstRTSPMediaClass *klass;
1942 GstRTSPMediaPrivate *priv;
1944 GstClockTime start, stop;
1945 GstSeekType start_type, stop_type;
1947 gint64 current_position;
1949 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1951 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1952 g_return_val_if_fail (range != NULL, FALSE);
1953 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1957 g_rec_mutex_lock (&priv->state_lock);
1958 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1961 /* Update the seekable state of the pipeline in case it changed */
1962 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
1963 /* TODO: Seeking for RECORD? */
1964 priv->seekable = FALSE;
1966 query = gst_query_new_seeking (GST_FORMAT_TIME);
1967 if (gst_element_query (priv->pipeline, query)) {
1972 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1973 priv->seekable = seekable;
1975 gst_query_unref (query);
1978 if (!priv->seekable)
1981 start_type = stop_type = GST_SEEK_TYPE_NONE;
1983 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1985 gst_rtsp_range_get_times (range, &start, &stop);
1987 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1988 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1989 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1990 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1992 current_position = -1;
1993 if (klass->query_position)
1994 klass->query_position (media, ¤t_position);
1995 GST_INFO ("current media position %" GST_TIME_FORMAT,
1996 GST_TIME_ARGS (current_position));
1998 if (start != GST_CLOCK_TIME_NONE)
1999 start_type = GST_SEEK_TYPE_SET;
2001 if (priv->range_stop == stop)
2002 stop = GST_CLOCK_TIME_NONE;
2003 else if (stop != GST_CLOCK_TIME_NONE)
2004 stop_type = GST_SEEK_TYPE_SET;
2006 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2009 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2010 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2012 /* depends on the current playing state of the pipeline. We might need to
2013 * queue this until we get EOS. */
2014 flags = GST_SEEK_FLAG_FLUSH;
2016 /* if range start was not supplied we must continue from current position.
2017 * but since we're doing a flushing seek, let us query the current position
2018 * so we end up at exactly the same position after the seek. */
2019 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2020 if (current_position == -1) {
2021 GST_WARNING ("current position unknown");
2023 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2024 GST_TIME_ARGS (current_position));
2025 start = current_position;
2026 start_type = GST_SEEK_TYPE_SET;
2027 flags |= GST_SEEK_FLAG_ACCURATE;
2030 /* only set keyframe flag when modifying start */
2031 if (start_type != GST_SEEK_TYPE_NONE)
2032 flags |= GST_SEEK_FLAG_KEY_UNIT;
2035 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2036 GST_DEBUG ("not seeking because no position change");
2039 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2041 media_streams_set_blocked (media, TRUE);
2043 /* FIXME, we only do forwards playback, no trick modes yet */
2044 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2045 flags, start_type, start, stop_type, stop);
2047 /* and block for the seek to complete */
2048 GST_INFO ("done seeking %d", res);
2052 g_rec_mutex_unlock (&priv->state_lock);
2054 /* wait until pipeline is prerolled again, this will also collect stats */
2055 if (!wait_preroll (media))
2056 goto preroll_failed;
2058 g_rec_mutex_lock (&priv->state_lock);
2059 GST_INFO ("prerolled again");
2062 GST_INFO ("no seek needed");
2065 g_rec_mutex_unlock (&priv->state_lock);
2072 g_rec_mutex_unlock (&priv->state_lock);
2073 GST_INFO ("media %p is not prepared", media);
2078 g_rec_mutex_unlock (&priv->state_lock);
2079 GST_INFO ("pipeline is not seekable");
2084 g_rec_mutex_unlock (&priv->state_lock);
2085 GST_WARNING ("conversion to npt not supported");
2090 g_rec_mutex_unlock (&priv->state_lock);
2091 GST_INFO ("seeking failed");
2092 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2097 GST_WARNING ("failed to preroll after seek");
2103 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2105 *blocked &= gst_rtsp_stream_is_blocking (stream);
2109 media_streams_blocking (GstRTSPMedia * media)
2111 gboolean blocking = TRUE;
2113 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2119 static GstStateChangeReturn
2120 set_state (GstRTSPMedia * media, GstState state)
2122 GstRTSPMediaPrivate *priv = media->priv;
2123 GstStateChangeReturn ret;
2125 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2127 ret = gst_element_set_state (priv->pipeline, state);
2132 static GstStateChangeReturn
2133 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2135 GstRTSPMediaPrivate *priv = media->priv;
2136 GstStateChangeReturn ret;
2138 GST_INFO ("set target state to %s for media %p",
2139 gst_element_state_get_name (state), media);
2140 priv->target_state = state;
2142 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2143 priv->target_state, NULL);
2146 ret = set_state (media, state);
2148 ret = GST_STATE_CHANGE_SUCCESS;
2153 /* called with state-lock */
2155 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2157 GstRTSPMediaPrivate *priv = media->priv;
2158 GstMessageType type;
2160 type = GST_MESSAGE_TYPE (message);
2163 case GST_MESSAGE_STATE_CHANGED:
2165 GstState old, new, pending;
2167 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2170 gst_message_parse_state_changed (message, &old, &new, &pending);
2172 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2173 gst_element_state_get_name (old), gst_element_state_get_name (new),
2174 gst_element_state_get_name (pending));
2175 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
2176 && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2177 GST_INFO ("%p: went to PAUSED, prepared now", media);
2178 collect_media_stats (media);
2180 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2181 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2186 case GST_MESSAGE_BUFFERING:
2190 gst_message_parse_buffering (message, &percent);
2192 /* no state management needed for live pipelines */
2196 if (percent == 100) {
2197 /* a 100% message means buffering is done */
2198 priv->buffering = FALSE;
2199 /* if the desired state is playing, go back */
2200 if (priv->target_state == GST_STATE_PLAYING) {
2201 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2202 set_state (media, GST_STATE_PLAYING);
2204 GST_INFO ("Buffering done");
2207 /* buffering busy */
2208 if (priv->buffering == FALSE) {
2209 if (priv->target_state == GST_STATE_PLAYING) {
2210 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2211 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2212 set_state (media, GST_STATE_PAUSED);
2214 GST_INFO ("Buffering ...");
2217 priv->buffering = TRUE;
2221 case GST_MESSAGE_LATENCY:
2223 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2226 case GST_MESSAGE_ERROR:
2231 gst_message_parse_error (message, &gerror, &debug);
2232 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2233 g_error_free (gerror);
2236 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2239 case GST_MESSAGE_WARNING:
2244 gst_message_parse_warning (message, &gerror, &debug);
2245 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2246 g_error_free (gerror);
2250 case GST_MESSAGE_ELEMENT:
2252 const GstStructure *s;
2254 s = gst_message_get_structure (message);
2255 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2256 GST_DEBUG ("media received blocking message");
2257 if (priv->blocked && media_streams_blocking (media)) {
2258 GST_DEBUG ("media is blocking");
2259 collect_media_stats (media);
2261 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2262 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2267 case GST_MESSAGE_STREAM_STATUS:
2269 case GST_MESSAGE_ASYNC_DONE:
2271 /* when we are dynamically adding pads, the addition of the udpsrc will
2272 * temporarily produce ASYNC_DONE messages. We have to ignore them and
2273 * wait for the final ASYNC_DONE after everything prerolled */
2274 GST_INFO ("%p: ignoring ASYNC_DONE", media);
2276 GST_INFO ("%p: got ASYNC_DONE", media);
2277 collect_media_stats (media);
2279 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2280 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2283 case GST_MESSAGE_EOS:
2284 GST_INFO ("%p: got EOS", media);
2286 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2287 GST_DEBUG ("shutting down after EOS");
2288 finish_unprepare (media);
2292 GST_INFO ("%p: got message type %d (%s)", media, type,
2293 gst_message_type_get_name (type));
2300 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2302 GstRTSPMediaPrivate *priv = media->priv;
2303 GstRTSPMediaClass *klass;
2306 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2308 g_rec_mutex_lock (&priv->state_lock);
2309 if (klass->handle_message)
2310 ret = klass->handle_message (media, message);
2313 g_rec_mutex_unlock (&priv->state_lock);
2319 watch_destroyed (GstRTSPMedia * media)
2321 GST_DEBUG_OBJECT (media, "source destroyed");
2322 g_object_unref (media);
2326 find_payload_element (GstElement * payloader)
2328 GstElement *pay = NULL;
2330 if (GST_IS_BIN (payloader)) {
2332 GValue item = { 0 };
2334 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2335 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2336 GstElement *element = (GstElement *) g_value_get_object (&item);
2337 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2341 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2345 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2346 pay = gst_object_ref (element);
2347 g_value_unset (&item);
2350 g_value_unset (&item);
2352 gst_iterator_free (iter);
2354 pay = g_object_ref (payloader);
2360 /* called from streaming threads */
2362 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2364 GstRTSPMediaPrivate *priv = media->priv;
2365 GstRTSPStream *stream;
2368 /* find the real payload element */
2369 pay = find_payload_element (element);
2370 stream = gst_rtsp_media_create_stream (media, pay, pad);
2371 gst_object_unref (pay);
2373 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2375 g_rec_mutex_lock (&priv->state_lock);
2376 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2379 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2381 /* we will be adding elements below that will cause ASYNC_DONE to be
2382 * posted in the bus. We want to ignore those messages until the
2383 * pipeline really prerolled. */
2384 priv->adding = TRUE;
2386 /* join the element in the PAUSED state because this callback is
2387 * called from the streaming thread and it is PAUSED */
2388 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2389 priv->rtpbin, GST_STATE_PAUSED)) {
2390 GST_WARNING ("failed to join bin element");
2393 priv->adding = FALSE;
2394 g_rec_mutex_unlock (&priv->state_lock);
2401 gst_rtsp_media_remove_stream (media, stream);
2402 g_rec_mutex_unlock (&priv->state_lock);
2403 GST_INFO ("ignore pad because we are not preparing");
2409 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2411 GstRTSPMediaPrivate *priv = media->priv;
2412 GstRTSPStream *stream;
2414 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2418 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2420 g_rec_mutex_lock (&priv->state_lock);
2421 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2422 g_rec_mutex_unlock (&priv->state_lock);
2424 gst_rtsp_media_remove_stream (media, stream);
2428 remove_fakesink (GstRTSPMediaPrivate * priv)
2430 GstElement *fakesink;
2432 g_mutex_lock (&priv->lock);
2433 if ((fakesink = priv->fakesink))
2434 gst_object_ref (fakesink);
2435 priv->fakesink = NULL;
2436 g_mutex_unlock (&priv->lock);
2439 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
2440 gst_element_set_state (fakesink, GST_STATE_NULL);
2441 gst_object_unref (fakesink);
2442 GST_INFO ("removed fakesink");
2447 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2449 GstRTSPMediaPrivate *priv = media->priv;
2451 GST_INFO ("no more pads");
2452 remove_fakesink (priv);
2455 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2457 struct _DynPaySignalHandlers
2459 gulong pad_added_handler;
2460 gulong pad_removed_handler;
2461 gulong no_more_pads_handler;
2465 start_preroll (GstRTSPMedia * media)
2467 GstRTSPMediaPrivate *priv = media->priv;
2468 GstStateChangeReturn ret;
2470 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2471 /* first go to PAUSED */
2472 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2475 case GST_STATE_CHANGE_SUCCESS:
2476 GST_INFO ("SUCCESS state change for media %p", media);
2477 priv->seekable = TRUE;
2479 case GST_STATE_CHANGE_ASYNC:
2480 GST_INFO ("ASYNC state change for media %p", media);
2481 priv->seekable = TRUE;
2483 case GST_STATE_CHANGE_NO_PREROLL:
2484 /* we need to go to PLAYING */
2485 GST_INFO ("NO_PREROLL state change: live media %p", media);
2486 /* FIXME we disable seeking for live streams for now. We should perform a
2487 * seeking query in preroll instead */
2488 priv->seekable = FALSE;
2489 priv->is_live = TRUE;
2490 if (!(priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
2491 /* start blocked to make sure nothing goes to the sink */
2492 media_streams_set_blocked (media, TRUE);
2494 ret = set_state (media, GST_STATE_PLAYING);
2495 if (ret == GST_STATE_CHANGE_FAILURE)
2498 case GST_STATE_CHANGE_FAILURE:
2506 GST_WARNING ("failed to preroll pipeline");
2512 wait_preroll (GstRTSPMedia * media)
2514 GstRTSPMediaStatus status;
2516 GST_DEBUG ("wait to preroll pipeline");
2518 /* wait until pipeline is prerolled */
2519 status = gst_rtsp_media_get_status (media);
2520 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2521 goto preroll_failed;
2527 GST_WARNING ("failed to preroll pipeline");
2533 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
2535 GstRTSPMediaPrivate *priv = media->priv;
2536 GstRTSPStream *stream = NULL;
2539 g_mutex_lock (&priv->lock);
2540 for (i = 0; i < priv->streams->len; i++) {
2541 stream = g_ptr_array_index (priv->streams, i);
2543 if (sessid == gst_rtsp_stream_get_index (stream))
2546 g_mutex_unlock (&priv->lock);
2548 return gst_rtsp_stream_request_aux_sender (stream, sessid);
2552 start_prepare (GstRTSPMedia * media)
2554 GstRTSPMediaPrivate *priv = media->priv;
2558 g_rec_mutex_lock (&priv->state_lock);
2559 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2560 goto no_longer_preparing;
2562 /* link streams we already have, other streams might appear when we have
2563 * dynamic elements */
2564 for (i = 0; i < priv->streams->len; i++) {
2565 GstRTSPStream *stream;
2567 stream = g_ptr_array_index (priv->streams, i);
2569 if (priv->rtx_time > 0) {
2570 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
2571 g_signal_connect (priv->rtpbin, "request-aux-sender",
2572 (GCallback) request_aux_sender, media);
2575 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2576 priv->rtpbin, GST_STATE_NULL)) {
2577 goto join_bin_failed;
2582 g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
2584 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2585 GstElement *elem = walk->data;
2586 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2588 GST_INFO ("adding callbacks for dynamic element %p", elem);
2590 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2591 (GCallback) pad_added_cb, media);
2592 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2593 (GCallback) pad_removed_cb, media);
2594 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2595 (GCallback) no_more_pads_cb, media);
2597 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2599 if (!priv->fakesink) {
2600 /* we add a fakesink here in order to make the state change async. We remove
2601 * the fakesink again in the no-more-pads callback. */
2602 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2603 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2607 if (!start_preroll (media))
2608 goto preroll_failed;
2610 g_rec_mutex_unlock (&priv->state_lock);
2614 no_longer_preparing:
2616 GST_INFO ("media is no longer preparing");
2617 g_rec_mutex_unlock (&priv->state_lock);
2622 GST_WARNING ("failed to join bin element");
2623 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2624 g_rec_mutex_unlock (&priv->state_lock);
2629 GST_WARNING ("failed to preroll pipeline");
2630 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2631 g_rec_mutex_unlock (&priv->state_lock);
2637 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2639 GstRTSPMediaPrivate *priv;
2640 GstRTSPMediaClass *klass;
2642 GMainContext *context;
2647 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2649 if (!klass->create_rtpbin)
2650 goto no_create_rtpbin;
2652 priv->rtpbin = klass->create_rtpbin (media);
2653 if (priv->rtpbin != NULL) {
2654 gboolean success = TRUE;
2656 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
2658 if (klass->setup_rtpbin)
2659 success = klass->setup_rtpbin (media, priv->rtpbin);
2661 if (success == FALSE) {
2662 gst_object_unref (priv->rtpbin);
2663 priv->rtpbin = NULL;
2666 if (priv->rtpbin == NULL)
2669 priv->thread = thread;
2670 context = (thread != NULL) ? (thread->context) : NULL;
2672 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2674 /* add the pipeline bus to our custom mainloop */
2675 priv->source = gst_bus_create_watch (bus);
2676 gst_object_unref (bus);
2678 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2679 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2681 priv->id = g_source_attach (priv->source, context);
2683 /* add stuff to the bin */
2684 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2686 /* do remainder in context */
2687 source = g_idle_source_new ();
2688 g_source_set_callback (source, (GSourceFunc) start_prepare,
2689 g_object_ref (media), (GDestroyNotify) g_object_unref);
2690 g_source_attach (source, context);
2691 g_source_unref (source);
2698 GST_ERROR ("no create_rtpbin function");
2699 g_critical ("no create_rtpbin vmethod function set");
2704 GST_WARNING ("no rtpbin element");
2705 g_warning ("failed to create element 'rtpbin', check your installation");
2711 * gst_rtsp_media_prepare:
2712 * @media: a #GstRTSPMedia
2713 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2714 * bus handler or %NULL
2716 * Prepare @media for streaming. This function will create the objects
2717 * to manage the streaming. A pipeline must have been set on @media with
2718 * gst_rtsp_media_take_pipeline().
2720 * It will preroll the pipeline and collect vital information about the streams
2721 * such as the duration.
2723 * Returns: %TRUE on success.
2726 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2728 GstRTSPMediaPrivate *priv;
2729 GstRTSPMediaClass *klass;
2731 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2735 g_rec_mutex_lock (&priv->state_lock);
2736 priv->prepare_count++;
2738 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2739 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2742 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2745 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2746 goto not_unprepared;
2748 if (!priv->reusable && priv->reused)
2751 GST_INFO ("preparing media %p", media);
2753 /* reset some variables */
2754 priv->is_live = FALSE;
2755 priv->seekable = FALSE;
2756 priv->buffering = FALSE;
2758 /* we're preparing now */
2759 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2761 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2762 if (klass->prepare) {
2763 if (!klass->prepare (media, thread))
2764 goto prepare_failed;
2768 g_rec_mutex_unlock (&priv->state_lock);
2770 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2771 * able to do this async so that we don't block the server thread. */
2772 if (!wait_preroll (media))
2773 goto preroll_failed;
2775 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2777 GST_INFO ("object %p is prerolled", media);
2784 /* we are not going to use the giving thread, so stop it. */
2786 gst_rtsp_thread_stop (thread);
2791 GST_LOG ("media %p was prepared", media);
2792 /* we are not going to use the giving thread, so stop it. */
2794 gst_rtsp_thread_stop (thread);
2795 g_rec_mutex_unlock (&priv->state_lock);
2801 /* we are not going to use the giving thread, so stop it. */
2803 gst_rtsp_thread_stop (thread);
2804 GST_WARNING ("media %p was not unprepared", media);
2805 priv->prepare_count--;
2806 g_rec_mutex_unlock (&priv->state_lock);
2811 /* we are not going to use the giving thread, so stop it. */
2813 gst_rtsp_thread_stop (thread);
2814 priv->prepare_count--;
2815 g_rec_mutex_unlock (&priv->state_lock);
2816 GST_WARNING ("can not reuse media %p", media);
2821 /* we are not going to use the giving thread, so stop it. */
2823 gst_rtsp_thread_stop (thread);
2824 priv->prepare_count--;
2825 g_rec_mutex_unlock (&priv->state_lock);
2826 GST_ERROR ("failed to prepare media");
2831 GST_WARNING ("failed to preroll pipeline");
2832 gst_rtsp_media_unprepare (media);
2837 /* must be called with state-lock */
2839 finish_unprepare (GstRTSPMedia * media)
2841 GstRTSPMediaPrivate *priv = media->priv;
2845 GST_DEBUG ("shutting down");
2847 /* release the lock on shutdown, otherwise pad_added_cb might try to
2848 * acquire the lock and then we deadlock */
2849 g_rec_mutex_unlock (&priv->state_lock);
2850 set_state (media, GST_STATE_NULL);
2851 g_rec_mutex_lock (&priv->state_lock);
2852 remove_fakesink (priv);
2854 for (i = 0; i < priv->streams->len; i++) {
2855 GstRTSPStream *stream;
2857 GST_INFO ("Removing elements of stream %d from pipeline", i);
2859 stream = g_ptr_array_index (priv->streams, i);
2861 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2864 /* remove the pad signal handlers */
2865 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2866 GstElement *elem = walk->data;
2867 DynPaySignalHandlers *handlers;
2870 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2871 g_assert (handlers != NULL);
2873 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2874 g_signal_handler_disconnect (G_OBJECT (elem),
2875 handlers->pad_removed_handler);
2876 g_signal_handler_disconnect (G_OBJECT (elem),
2877 handlers->no_more_pads_handler);
2879 g_slice_free (DynPaySignalHandlers, handlers);
2882 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2883 priv->rtpbin = NULL;
2886 gst_object_unref (priv->nettime);
2887 priv->nettime = NULL;
2889 priv->reused = TRUE;
2890 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
2892 /* when the media is not reusable, this will effectively unref the media and
2894 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2896 /* the source has the last ref to the media */
2898 GST_DEBUG ("destroy source");
2899 g_source_destroy (priv->source);
2900 g_source_unref (priv->source);
2903 GST_DEBUG ("stop thread");
2904 gst_rtsp_thread_stop (priv->thread);
2908 /* called with state-lock */
2910 default_unprepare (GstRTSPMedia * media)
2912 GstRTSPMediaPrivate *priv = media->priv;
2914 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
2916 if (priv->eos_shutdown) {
2917 GST_DEBUG ("sending EOS for shutdown");
2918 /* ref so that we don't disappear */
2919 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2920 /* we need to go to playing again for the EOS to propagate, normally in this
2921 * state, nothing is receiving data from us anymore so this is ok. */
2922 set_state (media, GST_STATE_PLAYING);
2924 finish_unprepare (media);
2930 * gst_rtsp_media_unprepare:
2931 * @media: a #GstRTSPMedia
2933 * Unprepare @media. After this call, the media should be prepared again before
2934 * it can be used again. If the media is set to be non-reusable, a new instance
2937 * Returns: %TRUE on success.
2940 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2942 GstRTSPMediaPrivate *priv;
2945 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2949 g_rec_mutex_lock (&priv->state_lock);
2950 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2951 goto was_unprepared;
2953 priv->prepare_count--;
2954 if (priv->prepare_count > 0)
2957 GST_INFO ("unprepare media %p", media);
2959 media_streams_set_blocked (media, FALSE);
2960 set_target_state (media, GST_STATE_NULL, FALSE);
2963 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2964 GstRTSPMediaClass *klass;
2966 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2967 if (klass->unprepare)
2968 success = klass->unprepare (media);
2970 finish_unprepare (media);
2972 g_rec_mutex_unlock (&priv->state_lock);
2978 g_rec_mutex_unlock (&priv->state_lock);
2979 GST_INFO ("media %p was already unprepared", media);
2984 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2985 g_rec_mutex_unlock (&priv->state_lock);
2990 /* should be called with state-lock */
2992 get_clock_unlocked (GstRTSPMedia * media)
2994 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2995 GST_DEBUG_OBJECT (media, "media was not prepared");
2998 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3002 * gst_rtsp_media_get_clock:
3003 * @media: a #GstRTSPMedia
3005 * Get the clock that is used by the pipeline in @media.
3007 * @media must be prepared before this method returns a valid clock object.
3009 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
3012 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3015 GstRTSPMediaPrivate *priv;
3017 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3021 g_rec_mutex_lock (&priv->state_lock);
3022 clock = get_clock_unlocked (media);
3023 g_rec_mutex_unlock (&priv->state_lock);
3029 * gst_rtsp_media_get_base_time:
3030 * @media: a #GstRTSPMedia
3032 * Get the base_time that is used by the pipeline in @media.
3034 * @media must be prepared before this method returns a valid base_time.
3036 * Returns: the base_time used by @media.
3039 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3041 GstClockTime result;
3042 GstRTSPMediaPrivate *priv;
3044 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3048 g_rec_mutex_lock (&priv->state_lock);
3049 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3052 result = gst_element_get_base_time (media->priv->pipeline);
3053 g_rec_mutex_unlock (&priv->state_lock);
3060 g_rec_mutex_unlock (&priv->state_lock);
3061 GST_DEBUG_OBJECT (media, "media was not prepared");
3062 return GST_CLOCK_TIME_NONE;
3067 * gst_rtsp_media_get_time_provider:
3068 * @media: a #GstRTSPMedia
3069 * @address: (allow-none): an address or %NULL
3070 * @port: a port or 0
3072 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3073 * will listen on @address and @port for client time requests.
3075 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3077 GstNetTimeProvider *
3078 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3081 GstRTSPMediaPrivate *priv;
3082 GstNetTimeProvider *provider = NULL;
3084 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3088 g_rec_mutex_lock (&priv->state_lock);
3089 if (priv->time_provider) {
3090 if ((provider = priv->nettime) == NULL) {
3093 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3094 provider = gst_net_time_provider_new (clock, address, port);
3095 gst_object_unref (clock);
3097 priv->nettime = provider;
3101 g_rec_mutex_unlock (&priv->state_lock);
3104 gst_object_ref (provider);
3110 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3112 return gst_rtsp_sdp_from_media (sdp, info, media);
3116 * gst_rtsp_media_setup_sdp:
3117 * @media: a #GstRTSPMedia
3118 * @sdp: (transfer none): a #GstSDPMessage
3119 * @info: (transfer none): a #GstSDPInfo
3121 * Add @media specific info to @sdp. @info is used to configure the connection
3122 * information in the SDP.
3124 * Returns: TRUE on success.
3127 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3130 GstRTSPMediaPrivate *priv;
3131 GstRTSPMediaClass *klass;
3134 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3135 g_return_val_if_fail (sdp != NULL, FALSE);
3136 g_return_val_if_fail (info != NULL, FALSE);
3140 g_rec_mutex_lock (&priv->state_lock);
3142 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3144 if (!klass->setup_sdp)
3147 res = klass->setup_sdp (media, sdp, info);
3149 g_rec_mutex_unlock (&priv->state_lock);
3156 g_rec_mutex_unlock (&priv->state_lock);
3157 GST_ERROR ("no setup_sdp function");
3158 g_critical ("no setup_sdp vmethod function set");
3163 static const gchar *
3164 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
3173 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
3176 if (sscanf (attr, "%d ", &val) != 1)
3185 #define PARSE_INT(p, del, res) \
3188 p = strstr (p, del); \
3198 #define PARSE_STRING(p, del, res) \
3201 p = strstr (p, del); \
3213 #define SKIP_SPACES(p) \
3214 while (*p && g_ascii_isspace (*p)) \
3219 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
3222 parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
3223 gint * rate, gchar ** params)
3227 p = (gchar *) rtpmap;
3229 PARSE_INT (p, " ", *payload);
3237 PARSE_STRING (p, "/", *name);
3238 if (*name == NULL) {
3239 GST_DEBUG ("no rate, name %s", p);
3240 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
3241 * streams seem to omit the rate. */
3248 p = strstr (p, "/");
3266 * Mapping of caps to and from SDP fields:
3268 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
3269 * a=framesize:<payload> <width>-<height>
3270 * a=fmtp:<payload> <param>[=<value>];...
3273 media_to_caps (gint pt, const GstSDPMedia * media)
3276 const gchar *rtpmap;
3278 const gchar *framesize;
3281 gchar *params = NULL;
3287 /* get and parse rtpmap */
3288 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
3291 ret = parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
3293 g_warning ("error parsing rtpmap, ignoring");
3297 /* dynamic payloads need rtpmap or we fail */
3298 if (rtpmap == NULL && pt >= 96)
3301 /* check if we have a rate, if not, we need to look up the rate from the
3302 * default rates based on the payload types. */
3304 const GstRTPPayloadInfo *info;
3306 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
3307 /* dynamic types, use media and encoding_name */
3308 tmp = g_ascii_strdown (media->media, -1);
3309 info = gst_rtp_payload_info_for_name (tmp, name);
3312 /* static types, use payload type */
3313 info = gst_rtp_payload_info_for_pt (pt);
3317 if ((rate = info->clock_rate) == 0)
3320 /* we fail if we cannot find one */
3325 tmp = g_ascii_strdown (media->media, -1);
3326 caps = gst_caps_new_simple ("application/x-unknown",
3327 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
3329 s = gst_caps_get_structure (caps, 0);
3331 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
3333 /* encoding name must be upper case */
3335 tmp = g_ascii_strup (name, -1);
3336 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
3340 /* params must be lower case */
3341 if (params != NULL) {
3342 tmp = g_ascii_strdown (params, -1);
3343 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
3347 /* parse optional fmtp: field */
3348 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
3354 /* p is now of the format <payload> <param>[=<value>];... */
3355 PARSE_INT (p, " ", payload);
3356 if (payload != -1 && payload == pt) {
3360 /* <param>[=<value>] are separated with ';' */
3361 pairs = g_strsplit (p, ";", 0);
3362 for (i = 0; pairs[i]; i++) {
3364 const gchar *val, *key;
3366 const gchar *reserved_keys[] =
3367 { "media", "payload", "clock-rate", "encoding-name",
3371 /* the key may not have a '=', the value can have other '='s */
3372 valpos = strstr (pairs[i], "=");
3374 /* we have a '=' and thus a value, remove the '=' with \0 */
3376 /* value is everything between '=' and ';'. We split the pairs at ;
3377 * boundaries so we can take the remainder of the value. Some servers
3378 * put spaces around the value which we strip off here. Alternatively
3379 * we could strip those spaces in the depayloaders should these spaces
3380 * actually carry any meaning in the future. */
3381 val = g_strstrip (valpos + 1);
3383 /* simple <param>;.. is translated into <param>=1;... */
3386 /* strip the key of spaces, convert key to lowercase but not the value. */
3387 key = g_strstrip (pairs[i]);
3389 /* skip keys from the fmtp, which we already use ourselves for the
3390 * caps. Some software is adding random things like clock-rate into
3391 * the fmtp, and we would otherwise here set a string-typed clock-rate
3392 * in the caps... and thus fail to create valid RTP caps
3394 for (j = 0; j < G_N_ELEMENTS (reserved_keys); j++) {
3395 if (g_ascii_strcasecmp (reserved_keys[j], key) == 0) {
3401 if (strlen (key) > 1) {
3402 tmp = g_ascii_strdown (key, -1);
3403 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
3411 /* parse framesize: field */
3412 if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
3415 /* p is now of the format <payload> <width>-<height> */
3416 p = (gchar *) framesize;
3418 PARSE_INT (p, " ", payload);
3419 if (payload != -1 && payload == pt) {
3420 gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
3428 g_warning ("rtpmap type not given for dynamic payload %d", pt);
3433 g_warning ("rate unknown for payload type %d", pt);
3439 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
3441 gboolean res = FALSE;
3444 GstMIKEYMessage *msg;
3445 const GstMIKEYPayload *payload;
3446 const gchar *srtp_cipher;
3447 const gchar *srtp_auth;
3453 p = orig_value = g_strdup (keymgmt);
3457 g_free (orig_value);
3461 PARSE_STRING (p, " ", kmpid);
3462 if (kmpid == NULL || !g_str_equal (kmpid, "mikey")) {
3463 g_free (orig_value);
3466 data = g_base64_decode (p, &size);
3468 g_free (orig_value); /* Don't need this any more */
3474 msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
3479 srtp_cipher = "aes-128-icm";
3480 srtp_auth = "hmac-sha1-80";
3482 /* check the Security policy if any */
3483 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
3484 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
3487 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
3490 len = gst_mikey_payload_sp_get_n_params (payload);
3491 for (i = 0; i < len; i++) {
3492 const GstMIKEYPayloadSPParam *param =
3493 gst_mikey_payload_sp_get_param (payload, i);
3495 switch (param->type) {
3496 case GST_MIKEY_SP_SRTP_ENC_ALG:
3497 switch (param->val[0]) {
3499 srtp_cipher = "null";
3503 srtp_cipher = "aes-128-icm";
3509 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
3510 switch (param->val[0]) {
3511 case AES_128_KEY_LEN:
3512 srtp_cipher = "aes-128-icm";
3514 case AES_256_KEY_LEN:
3515 srtp_cipher = "aes-256-icm";
3521 case GST_MIKEY_SP_SRTP_AUTH_ALG:
3522 switch (param->val[0]) {
3528 srtp_auth = "hmac-sha1-80";
3534 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
3535 switch (param->val[0]) {
3536 case HMAC_32_KEY_LEN:
3537 srtp_auth = "hmac-sha1-32";
3539 case HMAC_80_KEY_LEN:
3540 srtp_auth = "hmac-sha1-80";
3546 case GST_MIKEY_SP_SRTP_SRTP_ENC:
3548 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
3556 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
3559 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
3560 const GstMIKEYPayload *sub;
3561 GstMIKEYPayloadKeyData *pkd;
3564 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
3567 if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
3570 if (sub->type != GST_MIKEY_PT_KEY_DATA)
3573 pkd = (GstMIKEYPayloadKeyData *) sub;
3575 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
3577 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
3580 gst_caps_set_simple (caps,
3581 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
3582 "srtp-auth", G_TYPE_STRING, srtp_auth,
3583 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
3584 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
3588 gst_mikey_message_unref (msg);
3594 * Mapping SDP attributes to caps
3596 * prepend 'a-' to IANA registered sdp attributes names
3597 * (ie: not prefixed with 'x-') in order to avoid
3598 * collision with gstreamer standard caps properties names
3601 sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
3603 if (attributes->len > 0) {
3607 s = gst_caps_get_structure (caps, 0);
3609 for (i = 0; i < attributes->len; i++) {
3610 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
3611 gchar *tofree, *key;
3615 /* skip some of the attribute we already handle */
3616 if (!strcmp (key, "fmtp"))
3618 if (!strcmp (key, "rtpmap"))
3620 if (!strcmp (key, "control"))
3622 if (!strcmp (key, "range"))
3624 if (!strcmp (key, "framesize"))
3626 if (g_str_equal (key, "key-mgmt")) {
3627 parse_keymgmt (attr->value, caps);
3631 /* string must be valid UTF8 */
3632 if (!g_utf8_validate (attr->value, -1, NULL))
3635 if (!g_str_has_prefix (key, "x-"))
3636 tofree = key = g_strdup_printf ("a-%s", key);
3640 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
3641 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
3648 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3650 GstRTSPMediaPrivate *priv = media->priv;
3653 medias_len = gst_sdp_message_medias_len (sdp);
3654 if (medias_len != priv->streams->len) {
3655 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3656 priv->streams->len, medias_len);
3660 for (i = 0; i < medias_len; i++) {
3661 const gchar *proto, *media_type;
3662 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3663 GstRTSPStream *stream;
3664 gint j, formats_len;
3665 const gchar *control;
3666 GstRTSPProfile profile, profiles;
3668 stream = g_ptr_array_index (priv->streams, i);
3670 /* TODO: Should we do something with the other SDP information? */
3673 proto = gst_sdp_media_get_proto (sdp_media);
3674 if (proto == NULL) {
3675 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3679 if (g_str_equal (proto, "RTP/AVP")) {
3680 media_type = "application/x-rtp";
3681 profile = GST_RTSP_PROFILE_AVP;
3682 } else if (g_str_equal (proto, "RTP/SAVP")) {
3683 media_type = "application/x-srtp";
3684 profile = GST_RTSP_PROFILE_SAVP;
3685 } else if (g_str_equal (proto, "RTP/AVPF")) {
3686 media_type = "application/x-rtp";
3687 profile = GST_RTSP_PROFILE_AVPF;
3688 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3689 media_type = "application/x-srtp";
3690 profile = GST_RTSP_PROFILE_SAVPF;
3692 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3696 profiles = gst_rtsp_stream_get_profiles (stream);
3697 if ((profiles & profile) == 0) {
3698 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3702 formats_len = gst_sdp_media_formats_len (sdp_media);
3703 for (j = 0; j < formats_len; j++) {
3708 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3710 GST_DEBUG (" looking at %d pt: %d", j, pt);
3713 caps = media_to_caps (pt, sdp_media);
3715 GST_WARNING (" skipping pt %d without caps", pt);
3719 /* do some tweaks */
3720 GST_DEBUG ("mapping sdp session level attributes to caps");
3721 sdp_attributes_to_caps (sdp->attributes, caps);
3722 GST_DEBUG ("mapping sdp media level attributes to caps");
3723 sdp_attributes_to_caps (sdp_media->attributes, caps);
3725 s = gst_caps_get_structure (caps, 0);
3726 gst_structure_set_name (s, media_type);
3728 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3729 gst_caps_unref (caps);
3732 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3734 gst_rtsp_stream_set_control (stream, control);
3742 * gst_rtsp_media_handle_sdp:
3743 * @media: a #GstRTSPMedia
3744 * @sdp: (transfer none): a #GstSDPMessage
3746 * Configure an SDP on @media for receiving streams
3748 * Returns: TRUE on success.
3751 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3753 GstRTSPMediaPrivate *priv;
3754 GstRTSPMediaClass *klass;
3757 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3758 g_return_val_if_fail (sdp != NULL, FALSE);
3762 g_rec_mutex_lock (&priv->state_lock);
3764 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3766 if (!klass->handle_sdp)
3769 res = klass->handle_sdp (media, sdp);
3771 g_rec_mutex_unlock (&priv->state_lock);
3778 g_rec_mutex_unlock (&priv->state_lock);
3779 GST_ERROR ("no handle_sdp function");
3780 g_critical ("no handle_sdp vmethod function set");
3786 do_set_seqnum (GstRTSPStream * stream)
3789 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3790 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3793 /* call with state_lock */
3795 default_suspend (GstRTSPMedia * media)
3797 GstRTSPMediaPrivate *priv = media->priv;
3798 GstStateChangeReturn ret;
3799 gboolean unblock = FALSE;
3801 switch (priv->suspend_mode) {
3802 case GST_RTSP_SUSPEND_MODE_NONE:
3803 GST_DEBUG ("media %p no suspend", media);
3805 case GST_RTSP_SUSPEND_MODE_PAUSE:
3806 GST_DEBUG ("media %p suspend to PAUSED", media);
3807 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3808 if (ret == GST_STATE_CHANGE_FAILURE)
3812 case GST_RTSP_SUSPEND_MODE_RESET:
3813 GST_DEBUG ("media %p suspend to NULL", media);
3814 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3815 if (ret == GST_STATE_CHANGE_FAILURE)
3817 /* Because payloader needs to set the sequence number as
3818 * monotonic, we need to preserve the sequence number
3819 * after pause. (otherwise going from pause to play, which
3820 * is actually from NULL to PLAY will create a new sequence
3822 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3829 /* let the streams do the state changes freely, if any */
3831 media_streams_set_blocked (media, FALSE);
3838 GST_WARNING ("failed changing pipeline's state for media %p", media);
3844 * gst_rtsp_media_suspend:
3845 * @media: a #GstRTSPMedia
3847 * Suspend @media. The state of the pipeline managed by @media is set to
3848 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3849 * with gst_rtsp_media_unsuspend()
3851 * @media must be prepared with gst_rtsp_media_prepare();
3853 * Returns: %TRUE on success.
3856 gst_rtsp_media_suspend (GstRTSPMedia * media)
3858 GstRTSPMediaPrivate *priv = media->priv;
3859 GstRTSPMediaClass *klass;
3861 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3863 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3865 g_rec_mutex_lock (&priv->state_lock);
3866 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3869 /* don't attempt to suspend when something is busy */
3870 if (priv->n_active > 0)
3873 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3874 if (klass->suspend) {
3875 if (!klass->suspend (media))
3876 goto suspend_failed;
3879 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3881 g_rec_mutex_unlock (&priv->state_lock);
3888 g_rec_mutex_unlock (&priv->state_lock);
3889 GST_WARNING ("media %p was not prepared", media);
3894 g_rec_mutex_unlock (&priv->state_lock);
3895 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3896 GST_WARNING ("failed to suspend media %p", media);
3901 /* call with state_lock */
3903 default_unsuspend (GstRTSPMedia * media)
3905 GstRTSPMediaPrivate *priv = media->priv;
3907 switch (priv->suspend_mode) {
3908 case GST_RTSP_SUSPEND_MODE_NONE:
3909 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3911 case GST_RTSP_SUSPEND_MODE_PAUSE:
3912 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3914 case GST_RTSP_SUSPEND_MODE_RESET:
3916 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3917 if (!start_preroll (media))
3919 g_rec_mutex_unlock (&priv->state_lock);
3921 if (!wait_preroll (media))
3922 goto preroll_failed;
3924 g_rec_mutex_lock (&priv->state_lock);
3935 GST_WARNING ("failed to preroll pipeline");
3940 GST_WARNING ("failed to preroll pipeline");
3946 * gst_rtsp_media_unsuspend:
3947 * @media: a #GstRTSPMedia
3949 * Unsuspend @media if it was in a suspended state. This method does nothing
3950 * when the media was not in the suspended state.
3952 * Returns: %TRUE on success.
3955 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
3957 GstRTSPMediaPrivate *priv = media->priv;
3958 GstRTSPMediaClass *klass;
3960 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3962 g_rec_mutex_lock (&priv->state_lock);
3963 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3966 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3967 if (klass->unsuspend) {
3968 if (!klass->unsuspend (media))
3969 goto unsuspend_failed;
3973 g_rec_mutex_unlock (&priv->state_lock);
3980 g_rec_mutex_unlock (&priv->state_lock);
3981 GST_WARNING ("failed to unsuspend media %p", media);
3982 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3987 /* must be called with state-lock */
3989 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
3991 GstRTSPMediaPrivate *priv = media->priv;
3993 if (state == GST_STATE_NULL) {
3994 gst_rtsp_media_unprepare (media);
3996 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
3997 set_target_state (media, state, FALSE);
3998 /* when we are buffering, don't update the state yet, this will be done
3999 * when buffering finishes */
4000 if (priv->buffering) {
4001 GST_INFO ("Buffering busy, delay state change");
4003 if (state == GST_STATE_PLAYING)
4004 /* make sure pads are not blocking anymore when going to PLAYING */
4005 media_streams_set_blocked (media, FALSE);
4007 set_state (media, state);
4009 /* and suspend after pause */
4010 if (state == GST_STATE_PAUSED)
4011 gst_rtsp_media_suspend (media);
4017 * gst_rtsp_media_set_pipeline_state:
4018 * @media: a #GstRTSPMedia
4019 * @state: the target state of the pipeline
4021 * Set the state of the pipeline managed by @media to @state
4024 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4026 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4028 g_rec_mutex_lock (&media->priv->state_lock);
4029 media_set_pipeline_state_locked (media, state);
4030 g_rec_mutex_unlock (&media->priv->state_lock);
4034 * gst_rtsp_media_set_state:
4035 * @media: a #GstRTSPMedia
4036 * @state: the target state of the media
4037 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4038 * a #GPtrArray of #GstRTSPStreamTransport pointers
4040 * Set the state of @media to @state and for the transports in @transports.
4042 * @media must be prepared with gst_rtsp_media_prepare();
4044 * Returns: %TRUE on success.
4047 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4048 GPtrArray * transports)
4050 GstRTSPMediaPrivate *priv;
4052 gboolean activate, deactivate, do_state;
4055 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4056 g_return_val_if_fail (transports != NULL, FALSE);
4060 g_rec_mutex_lock (&priv->state_lock);
4061 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4063 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4064 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4067 /* NULL and READY are the same */
4068 if (state == GST_STATE_READY)
4069 state = GST_STATE_NULL;
4071 activate = deactivate = FALSE;
4073 GST_INFO ("going to state %s media %p, target state %s",
4074 gst_element_state_get_name (state), media,
4075 gst_element_state_get_name (priv->target_state));
4078 case GST_STATE_NULL:
4079 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4080 if (priv->target_state >= GST_STATE_PAUSED)
4083 case GST_STATE_PAUSED:
4084 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4085 if (priv->target_state == GST_STATE_PLAYING)
4088 case GST_STATE_PLAYING:
4089 /* we're going to PLAYING, activate */
4095 old_active = priv->n_active;
4097 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4098 activate, deactivate);
4099 for (i = 0; i < transports->len; i++) {
4100 GstRTSPStreamTransport *trans;
4102 /* we need a non-NULL entry in the array */
4103 trans = g_ptr_array_index (transports, i);
4108 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4110 } else if (deactivate) {
4111 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4116 /* we just activated the first media, do the playing state change */
4117 if (old_active == 0 && activate)
4119 /* if we have no more active media, do the downward state changes */
4120 else if (priv->n_active == 0)
4125 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4128 if (priv->target_state != state) {
4130 media_set_pipeline_state_locked (media, state);
4132 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4136 /* remember where we are */
4137 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4138 old_active != priv->n_active))
4139 collect_media_stats (media);
4141 g_rec_mutex_unlock (&priv->state_lock);
4148 GST_WARNING ("media %p was not prepared", media);
4149 g_rec_mutex_unlock (&priv->state_lock);
4154 GST_WARNING ("media %p in error status while changing to state %d",
4156 if (state == GST_STATE_NULL) {
4157 for (i = 0; i < transports->len; i++) {
4158 GstRTSPStreamTransport *trans;
4160 /* we need a non-NULL entry in the array */
4161 trans = g_ptr_array_index (transports, i);
4165 gst_rtsp_stream_transport_set_active (trans, FALSE);
4169 g_rec_mutex_unlock (&priv->state_lock);
4175 * gst_rtsp_media_set_transport_mode:
4176 * @media: a #GstRTSPMedia
4177 * @mode: the new value
4179 * Sets if the media pipeline can work in PLAY or RECORD mode
4182 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4183 GstRTSPTransportMode mode)
4185 GstRTSPMediaPrivate *priv;
4187 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4191 g_mutex_lock (&priv->lock);
4192 priv->transport_mode = mode;
4193 g_mutex_unlock (&priv->lock);
4197 * gst_rtsp_media_get_transport_mode:
4198 * @media: a #GstRTSPMedia
4200 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4202 * Returns: The transport mode.
4204 GstRTSPTransportMode
4205 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4207 GstRTSPMediaPrivate *priv;
4208 GstRTSPTransportMode res;
4210 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4214 g_mutex_lock (&priv->lock);
4215 res = priv->transport_mode;
4216 g_mutex_unlock (&priv->lock);