/* GStreamer * Copyright (C) 2008 Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #include #include #include #include #include "rtsp-media.h" #define GST_RTSP_MEDIA_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate)) struct _GstRTSPMediaPrivate { GMutex lock; GCond cond; /* protected by lock */ gboolean shared; gboolean reusable; GstRTSPLowerTrans protocols; gboolean reused; gboolean eos_shutdown; guint buffer_size; GstRTSPAuth *auth; GstRTSPAddressPool *pool; GstElement *element; GRecMutex state_lock; /* locking order: state lock, lock */ GPtrArray *streams; /* protected by lock */ GList *dynamic; /* protected by lock */ GstRTSPMediaStatus status; /* protected by lock */ gint prepare_count; gint n_active; gboolean adding; /* the pipeline for the media */ GstElement *pipeline; GstElement *fakesink; /* protected by lock */ GSource *source; guint id; gboolean is_live; gboolean seekable; gboolean buffering; GstState target_state; /* RTP session manager */ GstElement *rtpbin; /* the range of media */ GstRTSPTimeRange range; /* protected by lock */ GstClockTime range_start; GstClockTime range_stop; }; #define DEFAULT_SHARED FALSE #define DEFAULT_REUSABLE FALSE #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST #define DEFAULT_EOS_SHUTDOWN FALSE #define DEFAULT_BUFFER_SIZE 0x80000 /* define to dump received RTCP packets */ #undef DUMP_STATS enum { PROP_0, PROP_SHARED, PROP_REUSABLE, PROP_PROTOCOLS, PROP_EOS_SHUTDOWN, PROP_BUFFER_SIZE, PROP_ELEMENT, PROP_LAST }; enum { SIGNAL_NEW_STREAM, SIGNAL_PREPARED, SIGNAL_UNPREPARED, SIGNAL_NEW_STATE, SIGNAL_LAST }; GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug); #define GST_CAT_DEFAULT rtsp_media_debug static void gst_rtsp_media_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec); static void gst_rtsp_media_set_property (GObject * object, guint propid, const GValue * value, GParamSpec * pspec); static void gst_rtsp_media_finalize (GObject * obj); static gpointer do_loop (GstRTSPMediaClass * klass); static gboolean default_handle_message (GstRTSPMedia * media, GstMessage * message); static void finish_unprepare (GstRTSPMedia * media); static gboolean default_unprepare (GstRTSPMedia * media); static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 }; G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT); static void gst_rtsp_media_class_init (GstRTSPMediaClass * klass) { GObjectClass *gobject_class; g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate)); gobject_class = G_OBJECT_CLASS (klass); gobject_class->get_property = gst_rtsp_media_get_property; gobject_class->set_property = gst_rtsp_media_set_property; gobject_class->finalize = gst_rtsp_media_finalize; g_object_class_install_property (gobject_class, PROP_SHARED, g_param_spec_boolean ("shared", "Shared", "If this media pipeline can be shared", DEFAULT_SHARED, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_REUSABLE, g_param_spec_boolean ("reusable", "Reusable", "If this media pipeline can be reused after an unprepare", DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROTOCOLS, g_param_spec_flags ("protocols", "Protocols", "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS, DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN, g_param_spec_boolean ("eos-shutdown", "EOS Shutdown", "Send an EOS event to the pipeline before unpreparing", DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE, g_param_spec_uint ("buffer-size", "Buffer Size", "The kernel UDP buffer size to use", 0, G_MAXUINT, DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_ELEMENT, g_param_spec_object ("element", "The Element", "The GstBin to use for streaming the media", GST_TYPE_ELEMENT, G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE)); gst_rtsp_media_signals[SIGNAL_NEW_STREAM] = g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM); gst_rtsp_media_signals[SIGNAL_PREPARED] = g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); gst_rtsp_media_signals[SIGNAL_UNPREPARED] = g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); gst_rtsp_media_signals[SIGNAL_NEW_STATE] = g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL, g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT); klass->context = g_main_context_new (); klass->loop = g_main_loop_new (klass->context, TRUE); GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia"); klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass); klass->handle_message = default_handle_message; klass->unprepare = default_unprepare; } static void gst_rtsp_media_init (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media); media->priv = priv; priv->streams = g_ptr_array_new_with_free_func (g_object_unref); g_mutex_init (&priv->lock); g_cond_init (&priv->cond); g_rec_mutex_init (&priv->state_lock); priv->shared = DEFAULT_SHARED; priv->reusable = DEFAULT_REUSABLE; priv->protocols = DEFAULT_PROTOCOLS; priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN; priv->buffer_size = DEFAULT_BUFFER_SIZE; } static void gst_rtsp_media_finalize (GObject * obj) { GstRTSPMediaPrivate *priv; GstRTSPMedia *media; media = GST_RTSP_MEDIA (obj); priv = media->priv; GST_INFO ("finalize media %p", media); g_ptr_array_unref (priv->streams); g_list_free_full (priv->dynamic, gst_object_unref); if (priv->pipeline) gst_object_unref (priv->pipeline); gst_object_unref (priv->element); if (priv->auth) g_object_unref (priv->auth); if (priv->pool) g_object_unref (priv->pool); g_mutex_clear (&priv->lock); g_cond_clear (&priv->cond); g_rec_mutex_clear (&priv->state_lock); G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj); } static void gst_rtsp_media_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec) { GstRTSPMedia *media = GST_RTSP_MEDIA (object); switch (propid) { case PROP_ELEMENT: g_value_set_object (value, media->priv->element); break; case PROP_SHARED: g_value_set_boolean (value, gst_rtsp_media_is_shared (media)); break; case PROP_REUSABLE: g_value_set_boolean (value, gst_rtsp_media_is_reusable (media)); break; case PROP_PROTOCOLS: g_value_set_flags (value, gst_rtsp_media_get_protocols (media)); break; case PROP_EOS_SHUTDOWN: g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media)); break; case PROP_BUFFER_SIZE: g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } static void gst_rtsp_media_set_property (GObject * object, guint propid, const GValue * value, GParamSpec * pspec) { GstRTSPMedia *media = GST_RTSP_MEDIA (object); switch (propid) { case PROP_ELEMENT: media->priv->element = g_value_get_object (value); break; case PROP_SHARED: gst_rtsp_media_set_shared (media, g_value_get_boolean (value)); break; case PROP_REUSABLE: gst_rtsp_media_set_reusable (media, g_value_get_boolean (value)); break; case PROP_PROTOCOLS: gst_rtsp_media_set_protocols (media, g_value_get_flags (value)); break; case PROP_EOS_SHUTDOWN: gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value)); break; case PROP_BUFFER_SIZE: gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } static gpointer do_loop (GstRTSPMediaClass * klass) { GST_INFO ("enter mainloop"); g_main_loop_run (klass->loop); GST_INFO ("exit mainloop"); return NULL; } /* must be called with state lock */ static void collect_media_stats (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; gint64 position, duration; if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED && priv->status != GST_RTSP_MEDIA_STATUS_PREPARING) return; priv->range.unit = GST_RTSP_RANGE_NPT; GST_INFO ("collect media stats"); if (priv->is_live) { priv->range.min.type = GST_RTSP_TIME_NOW; priv->range.min.seconds = -1; priv->range_start = -1; priv->range.max.type = GST_RTSP_TIME_END; priv->range.max.seconds = -1; priv->range_stop = -1; } else { /* get the position */ if (!gst_element_query_position (priv->pipeline, GST_FORMAT_TIME, &position)) { GST_INFO ("position query failed"); position = 0; } /* get the duration */ if (!gst_element_query_duration (priv->pipeline, GST_FORMAT_TIME, &duration)) { GST_INFO ("duration query failed"); duration = -1; } GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration)); if (position == -1) { priv->range.min.type = GST_RTSP_TIME_NOW; priv->range.min.seconds = -1; priv->range_start = -1; } else { priv->range.min.type = GST_RTSP_TIME_SECONDS; priv->range.min.seconds = ((gdouble) position) / GST_SECOND; priv->range_start = position; } if (duration == -1) { priv->range.max.type = GST_RTSP_TIME_END; priv->range.max.seconds = -1; priv->range_stop = -1; } else { priv->range.max.type = GST_RTSP_TIME_SECONDS; priv->range.max.seconds = ((gdouble) duration) / GST_SECOND; priv->range_stop = duration; } } } /** * gst_rtsp_media_new: * @element: (transfer full): a #GstElement * * Create a new #GstRTSPMedia instance. @element is the bin element that * provides the different streams. The #GstRTSPMedia object contains the * element to produce RTP data for one or more related (audio/video/..) * streams. * * Ownership is taken of @element. * * Returns: a new #GstRTSPMedia object. */ GstRTSPMedia * gst_rtsp_media_new (GstElement * element) { GstRTSPMedia *result; g_return_val_if_fail (GST_IS_ELEMENT (element), NULL); result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL); return result; } /** * gst_rtsp_media_take_element: * @media: a #GstRTSPMedia * @pipeline: (transfer full): a #GstPipeline * * Set @pipeline as the #GstPipeline for @media. Ownership is * taken of @pipeline. */ void gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline) { GstRTSPMediaPrivate *priv; GstElement *old; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); g_return_if_fail (GST_IS_PIPELINE (pipeline)); priv = media->priv; g_mutex_lock (&priv->lock); old = priv->pipeline; priv->pipeline = GST_ELEMENT_CAST (pipeline); g_mutex_unlock (&priv->lock); if (old) gst_object_unref (old); gst_object_ref (priv->element); gst_bin_add (GST_BIN_CAST (pipeline), priv->element); } /** * gst_rtsp_media_set_shared: * @media: a #GstRTSPMedia * @shared: the new value * * Set or unset if the pipeline for @media can be shared will multiple clients. * When @shared is %TRUE, client requests for this media will share the media * pipeline. */ void gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared) { GstRTSPMediaPrivate *priv; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; g_mutex_lock (&priv->lock); priv->shared = shared; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_media_is_shared: * @media: a #GstRTSPMedia * * Check if the pipeline for @media can be shared between multiple clients. * * Returns: %TRUE if the media can be shared between clients. */ gboolean gst_rtsp_media_is_shared (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; gboolean res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); priv = media->priv; g_mutex_lock (&priv->lock); res = priv->shared; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_set_reusable: * @media: a #GstRTSPMedia * @reusable: the new value * * Set or unset if the pipeline for @media can be reused after the pipeline has * been unprepared. */ void gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable) { GstRTSPMediaPrivate *priv; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; g_mutex_lock (&priv->lock); priv->reusable = reusable; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_media_is_reusable: * @media: a #GstRTSPMedia * * Check if the pipeline for @media can be reused after an unprepare. * * Returns: %TRUE if the media can be reused */ gboolean gst_rtsp_media_is_reusable (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; gboolean res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); priv = media->priv; g_mutex_lock (&priv->lock); res = priv->reusable; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_set_protocols: * @media: a #GstRTSPMedia * @protocols: the new flags * * Configure the allowed lower transport for @media. */ void gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols) { GstRTSPMediaPrivate *priv; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; g_mutex_lock (&priv->lock); priv->protocols = protocols; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_media_get_protocols: * @media: a #GstRTSPMedia * * Get the allowed protocols of @media. * * Returns: a #GstRTSPLowerTrans */ GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; GstRTSPLowerTrans res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_LOWER_TRANS_UNKNOWN); priv = media->priv; g_mutex_lock (&priv->lock); res = priv->protocols; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_set_eos_shutdown: * @media: a #GstRTSPMedia * @eos_shutdown: the new value * * Set or unset if an EOS event will be sent to the pipeline for @media before * it is unprepared. */ void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown) { GstRTSPMediaPrivate *priv; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; g_mutex_lock (&priv->lock); priv->eos_shutdown = eos_shutdown; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_media_is_eos_shutdown: * @media: a #GstRTSPMedia * * Check if the pipeline for @media will send an EOS down the pipeline before * unpreparing. * * Returns: %TRUE if the media will send EOS before unpreparing. */ gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; gboolean res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); priv = media->priv; g_mutex_lock (&priv->lock); res = priv->eos_shutdown; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_set_buffer_size: * @media: a #GstRTSPMedia * @size: the new value * * Set the kernel UDP buffer size. */ void gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size) { GstRTSPMediaPrivate *priv; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); GST_LOG_OBJECT (media, "set buffer size %u", size); priv = media->priv; g_mutex_lock (&priv->lock); priv->buffer_size = size; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_media_get_buffer_size: * @media: a #GstRTSPMedia * * Get the kernel UDP buffer size. * * Returns: the kernel UDP buffer size. */ guint gst_rtsp_media_get_buffer_size (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; guint res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); priv = media->priv; g_mutex_unlock (&priv->lock); res = priv->buffer_size; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_set_auth: * @media: a #GstRTSPMedia * @auth: a #GstRTSPAuth * * configure @auth to be used as the authentication manager of @media. */ void gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth) { GstRTSPMediaPrivate *priv; GstRTSPAuth *old; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; GST_LOG_OBJECT (media, "set auth %p", auth); g_mutex_lock (&priv->lock); if ((old = priv->auth) != auth) priv->auth = auth ? g_object_ref (auth) : NULL; else old = NULL; g_mutex_unlock (&priv->lock); if (old) g_object_unref (old); } /** * gst_rtsp_media_get_auth: * @media: a #GstRTSPMedia * * Get the #GstRTSPAuth used as the authentication manager of @media. * * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after * usage. */ GstRTSPAuth * gst_rtsp_media_get_auth (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; GstRTSPAuth *result; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); priv = media->priv; g_mutex_lock (&priv->lock); if ((result = priv->auth)) g_object_ref (result); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_media_set_address_pool: * @media: a #GstRTSPMedia * @pool: a #GstRTSPAddressPool * * configure @pool to be used as the address pool of @media. */ void gst_rtsp_media_set_address_pool (GstRTSPMedia * media, GstRTSPAddressPool * pool) { GstRTSPMediaPrivate *priv; GstRTSPAddressPool *old; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; GST_LOG_OBJECT (media, "set address pool %p", pool); g_mutex_lock (&priv->lock); if ((old = priv->pool) != pool) priv->pool = pool ? g_object_ref (pool) : NULL; else old = NULL; g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool, pool); g_mutex_unlock (&priv->lock); if (old) g_object_unref (old); } /** * gst_rtsp_media_get_address_pool: * @media: a #GstRTSPMedia * * Get the #GstRTSPAddressPool used as the address pool of @media. * * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after * usage. */ GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; GstRTSPAddressPool *result; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); priv = media->priv; g_mutex_lock (&priv->lock); if ((result = priv->pool)) g_object_ref (result); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_media_collect_streams: * @media: a #GstRTSPMedia * * Find all payloader elements, they should be named pay%d in the * element of @media, and create #GstRTSPStreams for them. * * Collect all dynamic elements, named dynpay%d, and add them to * the list of dynamic elements. */ void gst_rtsp_media_collect_streams (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; GstElement *element, *elem; GstPad *pad; gint i; gboolean have_elem; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); priv = media->priv; element = priv->element; have_elem = TRUE; for (i = 0; have_elem; i++) { gchar *name; have_elem = FALSE; name = g_strdup_printf ("pay%d", i); if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) { GST_INFO ("found stream %d with payloader %p", i, elem); /* take the pad of the payloader */ pad = gst_element_get_static_pad (elem, "src"); /* create the stream */ gst_rtsp_media_create_stream (media, elem, pad); gst_object_unref (pad); gst_object_unref (elem); have_elem = TRUE; } g_free (name); name = g_strdup_printf ("dynpay%d", i); if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) { /* a stream that will dynamically create pads to provide RTP packets */ GST_INFO ("found dynamic element %d, %p", i, elem); g_mutex_lock (&priv->lock); priv->dynamic = g_list_prepend (priv->dynamic, elem); g_mutex_unlock (&priv->lock); have_elem = TRUE; } g_free (name); } } /** * gst_rtsp_media_create_stream: * @media: a #GstRTSPMedia * @payloader: a #GstElement * @srcpad: a source #GstPad * * Create a new stream in @media that provides RTP data on @srcpad. * @srcpad should be a pad of an element inside @media->element. * * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long * as @media exists. */ GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader, GstPad * pad) { GstRTSPMediaPrivate *priv; GstRTSPStream *stream; GstPad *srcpad; gchar *name; gint idx; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL); g_return_val_if_fail (GST_IS_PAD (pad), NULL); g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL); priv = media->priv; g_mutex_lock (&priv->lock); idx = priv->streams->len; GST_DEBUG ("media %p: creating stream with index %d", media, idx); name = g_strdup_printf ("src_%u", idx); srcpad = gst_ghost_pad_new (name, pad); gst_pad_set_active (srcpad, TRUE); gst_element_add_pad (priv->element, srcpad); g_free (name); stream = gst_rtsp_stream_new (idx, payloader, srcpad); if (priv->pool) gst_rtsp_stream_set_address_pool (stream, priv->pool); g_ptr_array_add (priv->streams, stream); g_mutex_unlock (&priv->lock); g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream, NULL); return stream; } /** * gst_rtsp_media_n_streams: * @media: a #GstRTSPMedia * * Get the number of streams in this media. * * Returns: The number of streams. */ guint gst_rtsp_media_n_streams (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; guint res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0); priv = media->priv; g_mutex_lock (&priv->lock); res = priv->streams->len; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_get_stream: * @media: a #GstRTSPMedia * @idx: the stream index * * Retrieve the stream with index @idx from @media. * * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with * that index did not exist. */ GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx) { GstRTSPMediaPrivate *priv; GstRTSPStream *res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); priv = media->priv; g_mutex_lock (&priv->lock); if (idx < priv->streams->len) res = g_ptr_array_index (priv->streams, idx); else res = NULL; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_media_get_range_string: * @media: a #GstRTSPMedia * @play: for the PLAY request * @unit: the unit to use for the string * * Get the current range as a string. @media must be prepared with * gst_rtsp_media_prepare (). * * Returns: The range as a string, g_free() after usage. */ gchar * gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play, GstRTSPRangeUnit unit) { GstRTSPMediaPrivate *priv; gchar *result; GstRTSPTimeRange range; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); priv = media->priv; g_rec_mutex_lock (&priv->state_lock); if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) goto not_prepared; g_mutex_lock (&priv->lock); /* make copy */ range = priv->range; if (!play && priv->n_active > 0) { range.min.type = GST_RTSP_TIME_NOW; range.min.seconds = -1; } g_mutex_unlock (&priv->lock); g_rec_mutex_unlock (&priv->state_lock); gst_rtsp_range_convert_units (&range, unit); result = gst_rtsp_range_to_string (&range); return result; /* ERRORS */ not_prepared: { GST_WARNING ("media %p was not prepared", media); g_rec_mutex_unlock (&priv->state_lock); return NULL; } } /** * gst_rtsp_media_seek: * @media: a #GstRTSPMedia * @range: a #GstRTSPTimeRange * * Seek the pipeline of @media to @range. @media must be prepared with * gst_rtsp_media_prepare(). * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range) { GstRTSPMediaPrivate *priv; GstSeekFlags flags; gboolean res; GstClockTime start, stop; GstSeekType start_type, stop_type; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); g_return_val_if_fail (range != NULL, FALSE); priv = media->priv; g_rec_mutex_lock (&priv->state_lock); if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) goto not_prepared; if (!priv->seekable) goto not_seekable; /* depends on the current playing state of the pipeline. We might need to * queue this until we get EOS. */ flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT; start_type = stop_type = GST_SEEK_TYPE_NONE; if (!gst_rtsp_range_get_times (range, &start, &stop)) goto not_supported; GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop)); GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop)); if (priv->range_start == start) start = GST_CLOCK_TIME_NONE; else if (start != GST_CLOCK_TIME_NONE) start_type = GST_SEEK_TYPE_SET; if (priv->range_stop == stop) stop = GST_CLOCK_TIME_NONE; else if (stop != GST_CLOCK_TIME_NONE) stop_type = GST_SEEK_TYPE_SET; if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) { GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop)); res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME, flags, start_type, start, stop_type, stop); /* and block for the seek to complete */ GST_INFO ("done seeking %d", res); gst_element_get_state (priv->pipeline, NULL, NULL, -1); GST_INFO ("prerolled again"); collect_media_stats (media); } else { GST_INFO ("no seek needed"); res = TRUE; } g_rec_mutex_unlock (&priv->state_lock); return res; /* ERRORS */ not_prepared: { g_rec_mutex_unlock (&priv->state_lock); GST_INFO ("media %p is not prepared", media); return FALSE; } not_seekable: { g_rec_mutex_unlock (&priv->state_lock); GST_INFO ("pipeline is not seekable"); return TRUE; } not_supported: { g_rec_mutex_unlock (&priv->state_lock); GST_WARNING ("seek unit %d not supported", range->unit); return FALSE; } } static void gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status) { GstRTSPMediaPrivate *priv = media->priv; g_mutex_lock (&priv->lock); priv->status = status; GST_DEBUG ("setting new status to %d", status); g_cond_broadcast (&priv->cond); g_mutex_unlock (&priv->lock); } /** * gst_rtsp_media_get_status: * @media: a #GstRTSPMedia * * Get the status of @media. When @media is busy preparing, this function waits * until @media is prepared or in error. * * Returns: the status of @media. */ GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; GstRTSPMediaStatus result; gint64 end_time; g_mutex_lock (&priv->lock); end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND; /* while we are preparing, wait */ while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) { GST_DEBUG ("waiting for status change"); if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) { GST_DEBUG ("timeout, assuming error status"); priv->status = GST_RTSP_MEDIA_STATUS_ERROR; } } /* could be success or error */ result = priv->status; GST_DEBUG ("got status %d", result); g_mutex_unlock (&priv->lock); return result; } /* called with state-lock */ static gboolean default_handle_message (GstRTSPMedia * media, GstMessage * message) { GstRTSPMediaPrivate *priv = media->priv; GstMessageType type; type = GST_MESSAGE_TYPE (message); switch (type) { case GST_MESSAGE_STATE_CHANGED: break; case GST_MESSAGE_BUFFERING: { gint percent; gst_message_parse_buffering (message, &percent); /* no state management needed for live pipelines */ if (priv->is_live) break; if (percent == 100) { /* a 100% message means buffering is done */ priv->buffering = FALSE; /* if the desired state is playing, go back */ if (priv->target_state == GST_STATE_PLAYING) { GST_INFO ("Buffering done, setting pipeline to PLAYING"); gst_element_set_state (priv->pipeline, GST_STATE_PLAYING); } else { GST_INFO ("Buffering done"); } } else { /* buffering busy */ if (priv->buffering == FALSE) { if (priv->target_state == GST_STATE_PLAYING) { /* we were not buffering but PLAYING, PAUSE the pipeline. */ GST_INFO ("Buffering, setting pipeline to PAUSED ..."); gst_element_set_state (priv->pipeline, GST_STATE_PAUSED); } else { GST_INFO ("Buffering ..."); } } priv->buffering = TRUE; } break; } case GST_MESSAGE_LATENCY: { gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline)); break; } case GST_MESSAGE_ERROR: { GError *gerror; gchar *debug; gst_message_parse_error (message, &gerror, &debug); GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug); g_error_free (gerror); g_free (debug); gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR); break; } case GST_MESSAGE_WARNING: { GError *gerror; gchar *debug; gst_message_parse_warning (message, &gerror, &debug); GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug); g_error_free (gerror); g_free (debug); break; } case GST_MESSAGE_ELEMENT: break; case GST_MESSAGE_STREAM_STATUS: break; case GST_MESSAGE_ASYNC_DONE: if (priv->adding) { /* when we are dynamically adding pads, the addition of the udpsrc will * temporarily produce ASYNC_DONE messages. We have to ignore them and * wait for the final ASYNC_DONE after everything prerolled */ GST_INFO ("%p: ignoring ASYNC_DONE", media); } else { GST_INFO ("%p: got ASYNC_DONE", media); collect_media_stats (media); if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED); } break; case GST_MESSAGE_EOS: GST_INFO ("%p: got EOS", media); if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) { GST_DEBUG ("shutting down after EOS"); finish_unprepare (media); } break; default: GST_INFO ("%p: got message type %d (%s)", media, type, gst_message_type_get_name (type)); break; } return TRUE; } static gboolean bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; GstRTSPMediaClass *klass; gboolean ret; klass = GST_RTSP_MEDIA_GET_CLASS (media); g_rec_mutex_lock (&priv->state_lock); if (klass->handle_message) ret = klass->handle_message (media, message); else ret = FALSE; g_rec_mutex_unlock (&priv->state_lock); return ret; } static void watch_destroyed (GstRTSPMedia * media) { GST_DEBUG_OBJECT (media, "source destroyed"); g_object_unref (media); } /* called from streaming threads */ static void pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; GstRTSPStream *stream; /* FIXME, element is likely not a payloader, find the payloader here */ stream = gst_rtsp_media_create_stream (media, element, pad); GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream); g_rec_mutex_lock (&priv->state_lock); /* we will be adding elements below that will cause ASYNC_DONE to be * posted in the bus. We want to ignore those messages until the * pipeline really prerolled. */ priv->adding = TRUE; /* join the element in the PAUSED state because this callback is * called from the streaming thread and it is PAUSED */ gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin, GST_STATE_PAUSED); priv->adding = FALSE; g_rec_mutex_unlock (&priv->state_lock); } static void no_more_pads_cb (GstElement * element, GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; GstElement *fakesink; g_mutex_lock (&priv->lock); GST_INFO ("no more pads"); if ((fakesink = priv->fakesink)) { gst_object_ref (fakesink); priv->fakesink = NULL; g_mutex_unlock (&priv->lock); gst_bin_remove (GST_BIN (priv->pipeline), fakesink); gst_element_set_state (fakesink, GST_STATE_NULL); gst_object_unref (fakesink); GST_INFO ("removed fakesink"); } } /** * gst_rtsp_media_prepare: * @media: a #GstRTSPMedia * * Prepare @media for streaming. This function will create the pipeline and * other objects to manage the streaming. * * It will preroll the pipeline and collect vital information about the streams * such as the duration. * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_prepare (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; GstStateChangeReturn ret; GstRTSPMediaStatus status; guint i; GstRTSPMediaClass *klass; GstBus *bus; GList *walk; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); priv = media->priv; g_rec_mutex_lock (&priv->state_lock); priv->prepare_count++; if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) goto was_prepared; if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) goto wait_status; if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED) goto not_unprepared; if (!priv->reusable && priv->reused) goto is_reused; priv->rtpbin = gst_element_factory_make ("rtpbin", NULL); if (priv->rtpbin == NULL) goto no_rtpbin; GST_INFO ("preparing media %p", media); /* reset some variables */ priv->is_live = FALSE; priv->seekable = FALSE; priv->buffering = FALSE; /* we're preparing now */ priv->status = GST_RTSP_MEDIA_STATUS_PREPARING; bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline)); /* add the pipeline bus to our custom mainloop */ priv->source = gst_bus_create_watch (bus); gst_object_unref (bus); g_source_set_callback (priv->source, (GSourceFunc) bus_message, g_object_ref (media), (GDestroyNotify) watch_destroyed); klass = GST_RTSP_MEDIA_GET_CLASS (media); priv->id = g_source_attach (priv->source, klass->context); /* add stuff to the bin */ gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin); /* link streams we already have, other streams might appear when we have * dynamic elements */ for (i = 0; i < priv->streams->len; i++) { GstRTSPStream *stream; stream = g_ptr_array_index (priv->streams, i); gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin, GST_STATE_NULL); } for (walk = priv->dynamic; walk; walk = g_list_next (walk)) { GstElement *elem = walk->data; GST_INFO ("adding callbacks for dynamic element %p", elem); g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media); g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media); /* we add a fakesink here in order to make the state change async. We remove * the fakesink again in the no-more-pads callback. */ priv->fakesink = gst_element_factory_make ("fakesink", "fakesink"); gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink); } GST_INFO ("setting pipeline to PAUSED for media %p", media); /* first go to PAUSED */ ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED); priv->target_state = GST_STATE_PAUSED; switch (ret) { case GST_STATE_CHANGE_SUCCESS: GST_INFO ("SUCCESS state change for media %p", media); priv->seekable = TRUE; break; case GST_STATE_CHANGE_ASYNC: GST_INFO ("ASYNC state change for media %p", media); priv->seekable = TRUE; break; case GST_STATE_CHANGE_NO_PREROLL: /* we need to go to PLAYING */ GST_INFO ("NO_PREROLL state change: live media %p", media); /* FIXME we disable seeking for live streams for now. We should perform a * seeking query in preroll instead */ priv->seekable = FALSE; priv->is_live = TRUE; ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING); if (ret == GST_STATE_CHANGE_FAILURE) goto state_failed; break; case GST_STATE_CHANGE_FAILURE: goto state_failed; } wait_status: g_rec_mutex_unlock (&priv->state_lock); /* now wait for all pads to be prerolled, FIXME, we should somehow be * able to do this async so that we don't block the server thread. */ status = gst_rtsp_media_get_status (media); if (status == GST_RTSP_MEDIA_STATUS_ERROR) goto state_failed; g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL); GST_INFO ("object %p is prerolled", media); return TRUE; /* OK */ was_prepared: { GST_LOG ("media %p was prepared", media); g_rec_mutex_unlock (&priv->state_lock); return TRUE; } /* ERRORS */ not_unprepared: { GST_WARNING ("media %p was not unprepared", media); priv->prepare_count--; g_rec_mutex_unlock (&priv->state_lock); return FALSE; } is_reused: { priv->prepare_count--; g_rec_mutex_unlock (&priv->state_lock); GST_WARNING ("can not reuse media %p", media); return FALSE; } no_rtpbin: { priv->prepare_count--; g_rec_mutex_unlock (&priv->state_lock); GST_WARNING ("no rtpbin element"); g_warning ("failed to create element 'rtpbin', check your installation"); return FALSE; } state_failed: { GST_WARNING ("failed to preroll pipeline"); gst_rtsp_media_unprepare (media); g_rec_mutex_unlock (&priv->state_lock); return FALSE; } } /* must be called with state-lock */ static void finish_unprepare (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; gint i; GST_DEBUG ("shutting down"); gst_element_set_state (priv->pipeline, GST_STATE_NULL); for (i = 0; i < priv->streams->len; i++) { GstRTSPStream *stream; GST_INFO ("Removing elements of stream %d from pipeline", i); stream = g_ptr_array_index (priv->streams, i); gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin); } g_ptr_array_set_size (priv->streams, 0); gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin); priv->rtpbin = NULL; gst_object_unref (priv->pipeline); priv->pipeline = NULL; priv->reused = TRUE; priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED; /* when the media is not reusable, this will effectively unref the media and * recreate it */ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL); /* the source has the last ref to the media */ if (priv->source) { GST_DEBUG ("destroy source"); g_source_destroy (priv->source); g_source_unref (priv->source); } } /* called with state-lock */ static gboolean default_unprepare (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv = media->priv; if (priv->eos_shutdown) { GST_DEBUG ("sending EOS for shutdown"); /* ref so that we don't disappear */ gst_element_send_event (priv->pipeline, gst_event_new_eos ()); /* we need to go to playing again for the EOS to propagate, normally in this * state, nothing is receiving data from us anymore so this is ok. */ gst_element_set_state (priv->pipeline, GST_STATE_PLAYING); priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING; } else { finish_unprepare (media); } return TRUE; } /** * gst_rtsp_media_unprepare: * @media: a #GstRTSPMedia * * Unprepare @media. After this call, the media should be prepared again before * it can be used again. If the media is set to be non-reusable, a new instance * must be created. * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_unprepare (GstRTSPMedia * media) { GstRTSPMediaPrivate *priv; gboolean success; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); priv = media->priv; g_rec_mutex_lock (&priv->state_lock); if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED) goto was_unprepared; priv->prepare_count--; if (priv->prepare_count > 0) goto is_busy; GST_INFO ("unprepare media %p", media); priv->target_state = GST_STATE_NULL; success = TRUE; if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) { GstRTSPMediaClass *klass; klass = GST_RTSP_MEDIA_GET_CLASS (media); if (klass->unprepare) success = klass->unprepare (media); } else { finish_unprepare (media); } g_rec_mutex_unlock (&priv->state_lock); return success; was_unprepared: { g_rec_mutex_unlock (&priv->state_lock); GST_INFO ("media %p was already unprepared", media); return TRUE; } is_busy: { GST_INFO ("media %p still prepared %d times", media, priv->prepare_count); g_rec_mutex_unlock (&priv->state_lock); return TRUE; } } /** * gst_rtsp_media_set_state: * @media: a #GstRTSPMedia * @state: the target state of the media * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers * * Set the state of @media to @state and for the transports in @transports. * * @media must be prepared with gst_rtsp_media_prepare(); * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state, GPtrArray * transports) { GstRTSPMediaPrivate *priv; gint i; gboolean activate, deactivate, do_state; gint old_active; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); g_return_val_if_fail (transports != NULL, FALSE); priv = media->priv; g_rec_mutex_lock (&priv->state_lock); if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) goto not_prepared; /* NULL and READY are the same */ if (state == GST_STATE_READY) state = GST_STATE_NULL; activate = deactivate = FALSE; GST_INFO ("going to state %s media %p", gst_element_state_get_name (state), media); switch (state) { case GST_STATE_NULL: case GST_STATE_PAUSED: /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */ if (priv->target_state == GST_STATE_PLAYING) deactivate = TRUE; break; case GST_STATE_PLAYING: /* we're going to PLAYING, activate */ activate = TRUE; break; default: break; } old_active = priv->n_active; for (i = 0; i < transports->len; i++) { GstRTSPStreamTransport *trans; /* we need a non-NULL entry in the array */ trans = g_ptr_array_index (transports, i); if (trans == NULL) continue; if (activate) { if (gst_rtsp_stream_transport_set_active (trans, TRUE)) priv->n_active++; } else if (deactivate) { if (gst_rtsp_stream_transport_set_active (trans, FALSE)) priv->n_active--; } } /* we just activated the first media, do the playing state change */ if (old_active == 0 && activate) do_state = TRUE; /* if we have no more active media, do the downward state changes */ else if (priv->n_active == 0) do_state = TRUE; else do_state = FALSE; GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active, media, do_state); if (priv->target_state != state) { if (do_state) { if (state == GST_STATE_NULL) { gst_rtsp_media_unprepare (media); } else { GST_INFO ("state %s media %p", gst_element_state_get_name (state), media); priv->target_state = state; /* when we are buffering, don't update the state yet, this will be done * when buffering finishes */ if (priv->buffering) { GST_INFO ("Buffering busy, delay state change"); } else { gst_element_set_state (priv->pipeline, state); } } } g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state, NULL); } /* remember where we are */ if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED || old_active != priv->n_active)) collect_media_stats (media); g_rec_mutex_unlock (&priv->state_lock); return TRUE; /* ERRORS */ not_prepared: { GST_WARNING ("media %p was not prepared", media); g_rec_mutex_unlock (&priv->state_lock); return FALSE; } }